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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@lists.bell-labs.com
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Subject: [SIP] Case sensitivity
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I believe we need to take a closer look at the annoying topic of
case-sensitivity (or not).

A rough summary is:

Method              CS
Header field name   CI
Any Accept-* token  CI  (includes compression, language, media type)
SDP encoding name   CI  (such as PCMU, L16, etc.)
URL                 mostly CS (except host)

Some additional issues:
(1) Parameters like ;level or ;q in Accept. Are ;LEVEL and ;Q the same
thing? RFC 2616 seems to indicate this:

3.7:

The type, subtype, and parameter attribute names are case-
   insensitive. Parameter values might or might not be case-sensitive,
   depending on the semantics of the parameter name.

2.1 Augmented BNF

"literal"
      Quotation marks surround literal text. Unless stated otherwise,
      the text is case-insensitive.

[This would apply to most enumerated parameters, like "hop" or "hide"
tokens in our case, or the "action" tokens in Contact or
Content-Disposition or the Priority values.]

My general impression is that HTTP/1.x seems to treat tokens not
enclosed in quotation marks as CI and anything in quotation marks as CS,
with URLs as a separate matter.

(2) HTTP-date (and, thus, rfc1123-date) is case sensitive.

Any other cases?
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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From sip-admin@lists.bell-labs.com  Mon May  1 10:43:27 2000
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From: James Yu <james.yu@neustar.com>
To: "'James Kempf'" <James.Kempf@Eng.Sun.COM>
Cc: sip@lists.bell-labs.com,
        "'SIGTRAN@STANDARDS.NORTELNETWORKS.COM'" <SIGTRAN@STANDARDS.NORTELNETWORKS.COM>
Subject: RE: [SIP] SIP Info Method and IS-41 Cellular Signalling Protocol
Date: Mon, 1 May 2000 09:41:52 -0500 
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James,

Your question can be addressed by sigtran - SS7 transport over IP.  There
are a few adaptation layers to choose including the newly proposed SUA (for
SCCP users such as IS-41).

James Yu

> -----Original Message-----
> From: James Kempf [mailto:James.Kempf@Eng.Sun.COM]
> Sent: Friday, April 14, 2000 5:59 PM
> To: sip@lists.bell-labs.com
> Cc: James.Kempf@Eng.Sun.COM
> Subject: [SIP] SIP Info Method and IS-41 Cellular Signalling Protocol
> 
> 
> I'm working on a design for replacing the IS-41 North 
> American cellular
> signalling protocol with IP protocols. I had a couple of 
> questions about
> how the SIP INFO method, or potentially another SIP method, might
> be used.
> 
> 1) In the introduction to draft-ietf-sip-info-method-03.txt, exchange
> of radio signal strength information is explicitly mentioned as
> an application of the SIP INFO method. The HandoffMeasurementRequest
> and HandoffMeasurementRequest2 messages in IS-41 are sent by 
> the anchor
> Mobile Switching Center (MSC) to a potential serving MSC to 
> obtain information 
> about the signal strength of a Mobile Station (MS) in order 
> to determine
> if the potential serving MSC is a better candidate for handling the
> mobile's call (called "hard handoff" in cellular jargon). 
> They could be 
> replaced with a SIP INFO method. There are three problems:
> 
> 	a) The radio information is always solicited by the anchor MSC,
> 	and from the ID, the SIP INFO method seems like it is 
> 	sent unsolicted. I suppose one could come up with two INFO
> 	methods, one to solicit the radio information and one to
> 	return it, but this seems like a hack somehow.
> 	
> 	b) In the ID, it explicitly states that the INFO method is 
> 	only applicable when there is a call in progress. Problem is,
> 	there isn't a call in progress between the anchor MSC and
> 	the serving MSC, that's why the anchor MSC is soliciting
> 	the radio quality information in the first place.
> 	
> 	c) The INFO method seems like it is meant to be sent between 
> 	SIP UAC's, while the HandoffMeasurementRequest(2) methods
> 	are primarily designed to be sent between network elements.
> 	A SIP replacement would probably be sent between the 
> last hop SIP
> 	proxies acting as Call Agents (CAs) for the call, which replaces
> 	the MSC in an all IP network.
> 
> 2) There are a bunch of IS-41 messages that involve transmitting
> feature or other information between the MS's Home Location
> Register (HLR) and the MS. These are initiated by either the serving
> MSC or the HLR. Examples are InformationDirective, which is used
> by the HLR to provide notifications to an idle MS, and FeatureRequest,
> which sends particular features to a potentially idle MS (features
> being things like messages or playing particular tones). Again, these
> seem like candidates to replace with the SIP INFO message, but 
> the potential lack of any call in progress would prohibit that.
> 
> The alternative I could see would be establishing a session in order
> to then send the information via the INFO method, but that seems like
> overkill. And in the case of HandoffMeasurementRequest(2), the
> message isn't intended for a SIP UAC anyway.
> 
> Maybe SIP is the wrong protocol to consider, since it is dedicated
> to setting up and maintaining sessions, and not to exchanging 
> information
> between network elements or between network elements and terminals?
> 
> Can anybody provide some guidance?
> 
> 		jak
> 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 


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I hesitate to wade into this because things tend to get somewhat
philosophical, but I think there are some important underlying design
issues. In any protocol (or grammar) design, there's syntax and semantics.
Unfortunately, they're not always cleanly separated in the specification. As
noted, ABNF is pretty good notation for syntax but really wasn't designed to
express semantics. Unfortunately, designers often try to capture semantics
in ABNF because it's a concise and formal way of doing so *when it works*.
But it's a slippery slope, and you often wind up with an ambiguous or
unusable syntactic specification as a result.

The URL parameter is a good example of this. It appears that the
"other-param" rule completely captures the syntax. The various other
parameter rules (transport-param, user-param, etc.) really capture semantics
rather than syntax. Now if all parameter names and values were enumerated,
these "enumerating" rules would capture both syntax and semantics and
"other-param" wouldn't be necessary. So presumably it was included to
account for yet unforeseen future parameters where the syntax needs to be
generically specified, but the semantics (including names and values) aren't
yet known. However in doing so, the semantic checking is eliminated;
"transport=fuddle" is perfectly valid syntax, so the apparent semantic
constraint implied by the transport-param rule has been lost. Is this a
transport parameter with an illegal value or a yet to be defined "other"
parameter. A syntactic or semantic violation, or no violation at all?

My own preference would be to define a core SIP syntax which is captured in
the ABNF. (ABNF does have a few problems, but that's another discussion.)
The current grammar and any future extensions should be parsable using this
syntax specification, because there's no attempt to define semantics.
Barring a formal semantic notation, all semantics is captured in the text as
is currently done, e.g., the text describing the  From header states:

The SIP-URL MUST NOT contain the "transport-param", "maddr-param",
"ttl-param", or "headers" elements. A server that receives a SIP-URL with
these elements removes them before further processing.

Note that the semantics includes not only legal parameter names and values
in that context, but what should occur when the "semantics" is violated.

So I think a more rigorous syntactic definition of SIP which is separate
from semantics would lead to a more concise core syntax, a more
understandable spec, and possibly better implementation designs, since there
would be less tendency to enforce semantic constraints in parsers which
would get re-written whenever an extension is added. I note that SIP is no
better or worse than many of the other IETF protocols in this respect.
(Computer language designers seem to have done a better job on this probably
because there is a much clearer separation of syntax and semantics in their
domain.)

Rick Workman
Nortel Networks
Ottawa

Henning Schulzrinne wrote:

> Maybe time to summarize and see if there's an issue left:
>
> - The host part of the URL cannot contain %, either as itself or as an
> escape mechanism:
>
> host          = hostname | IPv4address
> hostname      = *( domainlabel "." ) toplabel [ "." ]
> domainlabel   = alphanum | alphanum *( alphanum | "-" ) alphanum
> toplabel      = alpha | alpha *( alphanum | "-" ) alphanum
> alphanum      = alpha | digit
>
> - For "token" elements in URLs, this is a bit tricky, since we have two
> escape mechanisms that are intersecting:
>
>   (1) the normal token definition, i.e., any letter, digit, and a large
> number of symbols (-, #, $, &, %, ^, *, _, +, |, ~, ')
>
>   (2) the rules for URLs, where some of these characters (&, +) are
> reserved and thus need to be escaped if there's the possibility of
> confusion with a separator. As far as I can tell, neither of them are
> syntactically relevant for SIP URLs in this context (the & is only used
> to separate 'header' items), so neither of them would need to be
> escaped.
>
> Thus, it might be better to define
>
> other-param = pname [ "=" pvalue ]
>
> pname     = 1*tokenchar
> pvalue    = 1*tokenchar | quoted-string
> tokenchar = alphanum | escaped | "!" | "'" | "*" | "."
>             | "^" | "_" | "~" | "|" | "%" | "-" | "#" | "/" | ":" | "@"
> | "&" | "+" | "$"
>
> (since "," is not valid in token, I didn't include it. Any that I'm
> missing?)
>
> Here, escaped can only refer to the legal characters (alphanum and the
> others) to maintain "tokenness", but I can't express this in the BNF.
> However, in practice this is not a problem except for spec-writers since
> the pname and pvalue tokens have to be drawn from an enumeration, so
> that most character combinations are invalid simply because they are not
> valid parameter names.
>
> - To make matters slightly more confusing, we do have a different
> potential problem, namely the " in the quoted string. The <"> is a
> "delims" character and thus must be escaped (RFC 2396, pg. 9/10). One
> could argue that it only needs to be escaped if there's a possibility
> for confusion, e.g., on a web page or if the URL appears in a quoted
> string. However, either should be accepted by the parser.
>
> - In URLs, %hex hex (where allowed) is always interpreted as an escape
> mechanism. It is mechanically translated to the equivalent character,
> while maintaining its distinct identity as a non-separator (if it could
> be treated as such). Thus,
>
> foo%40bar@host.com
>
> can't just be translated into foo@bar@host.com and then be parsed, since
> the result would not be as expected. This is not fundamentally different
> from other escape mechanisms for delimiters, like \" or "".
>
> Thus, one should *first* split the URL into components and then
> translate those parts where escaping is possible (which is pretty much
> anywhere except in scheme and host).
>
> - As was pointed out, BNF specifies syntax. It cannot express certain
> things, including
>   . restrictions on ordering  (an | does not imply that the elements
> have to appear in order)
>   . any restriction on duplication
>   . the role of identifiers or restrictions on their names (e.g., "int
> int" may be legal in a BNF specification but the compiler may still
> complain)
>
>   For details, see for example, pg. 172 of Aho/Sethi/Ullman:
>
>   "Certain constraints on the input, such as the requirement that
> identifiers be declared before they are used, cannot be described by a
> context-free grammar."
>
> If there are places where the grammar needs to be tightened up, let me
> know.
>
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip



--------------19A162171A882F5FE670CDEC
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
&nbsp;
<br><tt>I hesitate to wade into this because things tend to get somewhat
philosophical, but I think there are some important underlying design issues.
In any protocol (or grammar) design, there's syntax and semantics. Unfortunately,
they're not always cleanly separated in the specification. As noted, ABNF
is pretty good notation for syntax but really wasn't designed to express
semantics. Unfortunately, designers often try to capture semantics in ABNF
because it's a concise and formal way of doing so *when it works*. But
it's a slippery slope, and you often wind up with an ambiguous or unusable
syntactic specification as a result.</tt><tt></tt>
<p><tt>The URL parameter is a good example of this. It appears that the
"other-param" rule completely captures the syntax. The various other parameter
rules (transport-param, user-param, etc.) really capture semantics rather
than syntax. Now if all parameter names and values were enumerated, these
"enumerating" rules would capture both syntax and semantics and "other-param"
wouldn't be necessary. So presumably it was included to account for yet
unforeseen future parameters where the syntax needs to be generically specified,
but the semantics (including names and values) aren't yet known. However
in doing so, the semantic checking is eliminated; "transport=fuddle" is
perfectly valid syntax, so the apparent semantic constraint implied by
the transport-param rule has been lost. Is this a transport parameter with
an illegal value or a yet to be defined "other" parameter. A syntactic
or semantic violation, or no violation at all?</tt><tt></tt>
<p><tt>My own preference would be to define a core SIP syntax which is
captured in the ABNF. (ABNF does have a few problems, but that's another
discussion.) The current grammar and any future extensions should be parsable
using this syntax specification, because there's no attempt to define semantics.
Barring a formal semantic notation, all semantics is captured in the text
as is currently done, e.g., the text describing the&nbsp; From header states:</tt><tt></tt>
<p><tt>The SIP-URL MUST NOT contain the "transport-param", "maddr-param",
"ttl-param", or "headers" elements. A server that receives a SIP-URL with
these elements removes them before further processing.</tt><tt></tt>
<p><tt>Note that the semantics includes not only legal parameter names
and values in that context, but what should occur when the "semantics"
is violated.</tt><tt></tt>
<p><tt>So I think a more rigorous syntactic definition of SIP which is
separate from semantics would lead to a more concise core syntax, a more
understandable spec, and possibly better implementation designs, since
there would be less tendency to enforce semantic constraints in parsers
which would get re-written whenever an extension is added. I note that
SIP is no better or worse than many of the other IETF protocols in this
respect. (Computer language designers seem to have done a better job on
this probably because there is a much clearer separation of syntax and
semantics in their domain.)</tt><tt></tt>
<p><tt>Rick Workman</tt>
<br><tt>Nortel Networks</tt>
<br><tt>Ottawa</tt><tt></tt>
<p><tt>Henning Schulzrinne wrote:</tt>
<blockquote TYPE=CITE><tt>Maybe time to summarize and see if there's an
issue left:</tt><tt></tt>
<p><tt>- The host part of the URL cannot contain %, either as itself or
as an</tt>
<br><tt>escape mechanism:</tt><tt></tt>
<p><tt>host&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = hostname
| IPv4address</tt>
<br><tt>hostname&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = *( domainlabel "." ) toplabel
[ "." ]</tt>
<br><tt>domainlabel&nbsp;&nbsp; = alphanum | alphanum *( alphanum | "-"
) alphanum</tt>
<br><tt>toplabel&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = alpha | alpha *( alphanum
| "-" ) alphanum</tt>
<br><tt>alphanum&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = alpha | digit</tt><tt></tt>
<p><tt>- For "token" elements in URLs, this is a bit tricky, since we have
two</tt>
<br><tt>escape mechanisms that are intersecting:</tt><tt></tt>
<p><tt>&nbsp; (1) the normal token definition, i.e., any letter, digit,
and a large</tt>
<br><tt>number of symbols (-, #, $, &amp;, %, ^, *, _, +, |, ~, ')</tt><tt></tt>
<p><tt>&nbsp; (2) the rules for URLs, where some of these characters (&amp;,
+) are</tt>
<br><tt>reserved and thus need to be escaped if there's the possibility
of</tt>
<br><tt>confusion with a separator. As far as I can tell, neither of them
are</tt>
<br><tt>syntactically relevant for SIP URLs in this context (the &amp;
is only used</tt>
<br><tt>to separate 'header' items), so neither of them would need to be</tt>
<br><tt>escaped.</tt><tt></tt>
<p><tt>Thus, it might be better to define</tt><tt></tt>
<p><tt>other-param = pname [ "=" pvalue ]</tt><tt></tt>
<p><tt>pname&nbsp;&nbsp;&nbsp;&nbsp; = 1*tokenchar</tt>
<br><tt>pvalue&nbsp;&nbsp;&nbsp; = 1*tokenchar | quoted-string</tt>
<br><tt>tokenchar = alphanum | escaped | "!" | "'" | "*" | "."</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
| "^" | "_" | "~" | "|" | "%" | "-" | "#" | "/" | ":" | "@"</tt>
<br><tt>| "&amp;" | "+" | "$"</tt><tt></tt>
<p><tt>(since "," is not valid in token, I didn't include it. Any that
I'm</tt>
<br><tt>missing?)</tt><tt></tt>
<p><tt>Here, escaped can only refer to the legal characters (alphanum and
the</tt>
<br><tt>others) to maintain "tokenness", but I can't express this in the
BNF.</tt>
<br><tt>However, in practice this is not a problem except for spec-writers
since</tt>
<br><tt>the pname and pvalue tokens have to be drawn from an enumeration,
so</tt>
<br><tt>that most character combinations are invalid simply because they
are not</tt>
<br><tt>valid parameter names.</tt><tt></tt>
<p><tt>- To make matters slightly more confusing, we do have a different</tt>
<br><tt>potential problem, namely the " in the quoted string. The &lt;">
is a</tt>
<br><tt>"delims" character and thus must be escaped (RFC 2396, pg. 9/10).
One</tt>
<br><tt>could argue that it only needs to be escaped if there's a possibility</tt>
<br><tt>for confusion, e.g., on a web page or if the URL appears in a quoted</tt>
<br><tt>string. However, either should be accepted by the parser.</tt><tt></tt>
<p><tt>- In URLs, %hex hex (where allowed) is always interpreted as an
escape</tt>
<br><tt>mechanism. It is mechanically translated to the equivalent character,</tt>
<br><tt>while maintaining its distinct identity as a non-separator (if
it could</tt>
<br><tt>be treated as such). Thus,</tt><tt></tt>
<p><tt>foo%40bar@host.com</tt><tt></tt>
<p><tt>can't just be translated into foo@bar@host.com and then be parsed,
since</tt>
<br><tt>the result would not be as expected. This is not fundamentally
different</tt>
<br><tt>from other escape mechanisms for delimiters, like \" or "".</tt><tt></tt>
<p><tt>Thus, one should *first* split the URL into components and then</tt>
<br><tt>translate those parts where escaping is possible (which is pretty
much</tt>
<br><tt>anywhere except in scheme and host).</tt><tt></tt>
<p><tt>- As was pointed out, BNF specifies syntax. It cannot express certain</tt>
<br><tt>things, including</tt>
<br><tt>&nbsp; . restrictions on ordering&nbsp; (an | does not imply that
the elements</tt>
<br><tt>have to appear in order)</tt>
<br><tt>&nbsp; . any restriction on duplication</tt>
<br><tt>&nbsp; . the role of identifiers or restrictions on their names
(e.g., "int</tt>
<br><tt>int" may be legal in a BNF specification but the compiler may still</tt>
<br><tt>complain)</tt><tt></tt>
<p><tt>&nbsp; For details, see for example, pg. 172 of Aho/Sethi/Ullman:</tt><tt></tt>
<p><tt>&nbsp; "Certain constraints on the input, such as the requirement
that</tt>
<br><tt>identifiers be declared before they are used, cannot be described
by a</tt>
<br><tt>context-free grammar."</tt><tt></tt>
<p><tt>If there are places where the grammar needs to be tightened up,
let me</tt>
<br><tt>know.</tt><tt></tt>
<p><tt>--</tt>
<br><tt>Henning Schulzrinne&nbsp;&nbsp; <a href="http://www.cs.columbia.edu/~hgs">http://www.cs.columbia.edu/~hgs</a></tt><tt></tt>
<p><tt>_______________________________________________</tt>
<br><tt>SIP mailing list</tt>
<br><tt>SIP@lists.bell-labs.com</tt>
<br><tt><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></tt></blockquote>
<tt></tt>
<br>&nbsp;</html>

--------------19A162171A882F5FE670CDEC--



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Date: Mon, 01 May 2000 09:11:15 -0700
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [SIP] mwi in SIP: supplying waiting message lists
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Jonathan,

I took a good look at IMAP, POP, and VPIM for message waiting.  The models 
were too different to use, as Henning begins to hint.  The only thing that 
survived serious scrutiny was the mailbox status definitions used in 
IMAP.  I proposed an XML-based representation of mailbox summary inside 
Cisco.  Ilya proposed a more-simple text-based representation of the same 
information, which is what you see below.

We felt that the UA should have access to a summary so it could display 
things like:

         "You have 2 new, and 5 old voice messages; and 1 old fax message."

It would be very nice to have some proposed solution for message waiting.

thanks,
-rohan

message status borrowed from IMAP:
> > U = UNTOUCHED
> > S = SKIPPED
> > F = FLAGGED
> > R = READ
> > A = ANSWERED
> > D = DELETED

thanks,
-rohan



At 11:53 PM 4/27/00 , Jonathan Rosenberg wrote:
>Message waiting has come up numerous times. I agree we should have a
>solution based on a subscribe/notify method. Just to note, I do believe
>there are other ways to do this that can make use of other existing
>protocols. For systems that use email-based voicemail storage (like
>VPIM), IMAP may actually be the appropriate protocol for message
>waiting. I accept that IMAP is gross overkill for systems that just want
>voicemail only message waiting lamps, so something a little more focused
>would be useful.
>
>As pointed out in another thread, the described mechanism of enumerating
>events in SUBSCRIBE is too restrictive and not compatible with pint.
>Rather than attacking these bit at a time, I think it would be
>tremendously valuable to list the general set of events and state
>associated with SIP communications systems that one might wish to
>subscribe to. This will help us figure out the set of things we might
>like to build in to some set of "sip presence subscription documents",
>which are carried in SUBSCRIBE messages to indicate what is being
>subscribed to.
>
>-Jonathan R.
>
>Ilya Slain wrote:
> >
> > There has recently been some discussion on the way to
> > do message waiting indication (mwi) using SUBSCRIBE/
> > NOTIFY SIP extensions.  However, I am not aware of
> > any proposals to for specifying waiting message list
> > summary, supplied in mwi NOTIFY message.  In other
> > words, there is no proposed way to supply the summary
> > of messages being available in the voice mail system.
> >
> > So I would like to submit a brief sample proposal to
> > do
> > message waiting indication with message summary using
> > SIP
> > SUBSCRIBE/NOTIFY.  This is based on the recent
> > SUBSCSRIBE/
> > NOTIFY IETF draft:
> > 
> http://search.ietf.org/internet-drafts/draft-roach-sip-subscribe-notify-00.txt
> >
> > The main purpose of this proposal is to start the
> > discussion
> > on the topic with the aim of eventually submitting an
> > Inf. Draft.
> >
> > 1. Subscription
> > ----------------
> > Event field specifies "mwi" field (message waiting
> > indication).
> > There is no message body.
> >
> > SUBSCRIBE ...
> > To:             ...
> > From:           ...
> > Call-ID:        ...
> > CSeq:           ...
> > Contact:        ...
> > Expires:        ...
> > Event:          mwi
> > Content-Length: 0
> >
> > 2. Notification
> > -----------------
> > Event field specifies "mwi".
> > Message body specifies the message summary.
> >
> > NOTIFY
> > To:             ...
> > From:           ...
> > Call-ID:        ...
> > CSeq:           ...
> > Contact:        ...
> > Expires:        ...
> > Event:          mwi
> > Content-Type:   application/um
> > Content-Length: 148
> >
> > VOICEMAIL: <TOTAL_MESSAGES>; U:n1/n2; S:n1/n2;
> > F:n1/n2; R:n1/n2; A:n1/n2; D:n1/n2
> > EMAIL:     <TOTAL_MESSAGES>; U:n1/n2; S:n1/n2;
> > F:n1/n2; R:n1/n2; A:n1/n2; D:n1/n2
> > FAX:       <TOTAL_MESSAGES>; U:n1/n2; S:n1/n2;
> > F:n1/n2; R:n1/n2; A:n1/n2; D:n1/n2
> > VIDEO:     <TOTAL_MESSAGES>; U:n1/n2; S:n1/n2;
> > F:n1/n2; R:n1/n2; A:n1/n2; D:n1/n2
> >
> > where TOTAL_MESSAGES, n1, n2 are integers
> > U = UNTOUCHED
> > S = SKIPPED
> > F = FLAGGED
> > R = READ
> > A = ANSWERED
> > D = DELETED
> >
> > For example:
> > voicemail: 10; U:3/1; S:1/0; F:0/0; R:5/2; A:1/0;
> > D:0/0
> > where U: 3/1  indicates that there are 3 untouched
> > messages, 1 of which is urgent.
> >
> > More formal grammar for the application/um message
> > body:
> >
> > application_um_message_body = message_summary_list
> > message_summary_list = message_summary "\n"
> > message_summary_list | NULL
> > message_summary = message_type ": " total_messages ";
> > " typed_summary_list
> > message_type = "VOICEMAIL" | "EMAIL" | "FAX" | "VIDEO"
> > total_messages = INTEGER
> > typed_summary_list = typed_summary ";"
> > typed_summary_list | NULL
> > typed_summary = type ":" number_of_messages "/"
> > number_of_urgent_messages
> > type = "U" | "S" | "F" | "R" | "A" | "D"
> > number_of_messages = INTEGER
> > number_of_urgent_messages = INTEGER
> >
> > 
> ______________________________________________________________________________
> >
> > Ilya Slain                       ||        ||
> >   Cisco Systems, Inc.
> > Technology Center               .||.      .||.
> >            Building 9
> >                                .||||.    .||||.
> > 170 West Tasman Drive
> > Tel:    408.527.9446       ...||||||||..||||||||...
> >   San Jose, CA  95134
> > Fax:    408.527.1221       C i s c o  S y s t e m s
> > E-Mail: islain@cisco.com
> > 
> ____________________________________________________________________________
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Send online invitations with Yahoo! Invites.
> > http://invites.yahoo.com
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
>
>--
>Jonathan D. Rosenberg                       72 Eagle Rock Ave.
>Chief Scientist                             First Floor
>dynamicsoft                                 East Hanover, NJ 07936
>jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
>http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
>http://www.dynamicsoft.com
>
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip



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From sip-admin@lists.bell-labs.com  Mon May  1 13:13:50 2000
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Date: Mon, 01 May 2000 10:11:45 -0700
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Fwd: [SIP] Sip URL and escaped characters. RFC 2543
  Section 2]
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Sip Mail List <sip@lists.bell-labs.com>
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Hi,

I just want to make sure that we can handle/describe the escaping of a 
nested monster such as this:  (sorrry for the wrapping at hyphens)

<a 
href="<sip:userA.butler@sip.cisco.com:5060;method=TRANSFER?Transfer-To=<sip: 
+14083079999@sip.cisco.com:5060;user=phone?Request-Disposition=no-fork&Accep 
t-Contact=*;mobility="mobile",*;media="audio">&Request-Disposition=parallel>">

this would be a SIP URI inside an HTML page, that asks userA's "Butler" 
(wherever he/she might be) to call (presumably userA's) cell phone.

this particular example may be a bit contrived, but i can come up with 
legitimate uses of this kind of functionality.

thanks,
-rohan

At 11:37 AM 4/29/00 , Henning Schulzrinne wrote:
>Maybe time to summarize and see if there's an issue left:
>
>- The host part of the URL cannot contain %, either as itself or as an
>escape mechanism:
>
>host          = hostname | IPv4address
>hostname      = *( domainlabel "." ) toplabel [ "." ]
>domainlabel   = alphanum | alphanum *( alphanum | "-" ) alphanum
>toplabel      = alpha | alpha *( alphanum | "-" ) alphanum
>alphanum      = alpha | digit
>
>- For "token" elements in URLs, this is a bit tricky, since we have two
>escape mechanisms that are intersecting:
>
>   (1) the normal token definition, i.e., any letter, digit, and a large
>number of symbols (-, #, $, &, %, ^, *, _, +, |, ~, ')
>
>   (2) the rules for URLs, where some of these characters (&, +) are
>reserved and thus need to be escaped if there's the possibility of
>confusion with a separator. As far as I can tell, neither of them are
>syntactically relevant for SIP URLs in this context (the & is only used
>to separate 'header' items), so neither of them would need to be
>escaped.
>
>Thus, it might be better to define
>
>other-param = pname [ "=" pvalue ]
>
>pname     = 1*tokenchar
>pvalue    = 1*tokenchar | quoted-string
>tokenchar = alphanum | escaped | "!" | "'" | "*" | "."
>             | "^" | "_" | "~" | "|" | "%" | "-" | "#" | "/" | ":" | "@"
>| "&" | "+" | "$"
>
>(since "," is not valid in token, I didn't include it. Any that I'm
>missing?)
>
>Here, escaped can only refer to the legal characters (alphanum and the
>others) to maintain "tokenness", but I can't express this in the BNF.
>However, in practice this is not a problem except for spec-writers since
>the pname and pvalue tokens have to be drawn from an enumeration, so
>that most character combinations are invalid simply because they are not
>valid parameter names.
>
>- To make matters slightly more confusing, we do have a different
>potential problem, namely the " in the quoted string. The <"> is a
>"delims" character and thus must be escaped (RFC 2396, pg. 9/10). One
>could argue that it only needs to be escaped if there's a possibility
>for confusion, e.g., on a web page or if the URL appears in a quoted
>string. However, either should be accepted by the parser.
>
>
>- In URLs, %hex hex (where allowed) is always interpreted as an escape
>mechanism. It is mechanically translated to the equivalent character,
>while maintaining its distinct identity as a non-separator (if it could
>be treated as such). Thus,
>
>foo%40bar@host.com
>
>can't just be translated into foo@bar@host.com and then be parsed, since
>the result would not be as expected. This is not fundamentally different
>from other escape mechanisms for delimiters, like \" or "".
>
>Thus, one should *first* split the URL into components and then
>translate those parts where escaping is possible (which is pretty much
>anywhere except in scheme and host).
>
>- As was pointed out, BNF specifies syntax. It cannot express certain
>things, including
>   . restrictions on ordering  (an | does not imply that the elements
>have to appear in order)
>   . any restriction on duplication
>   . the role of identifiers or restrictions on their names (e.g., "int
>int" may be legal in a BNF specification but the compiler may still
>complain)
>
>   For details, see for example, pg. 172 of Aho/Sethi/Ullman:
>
>   "Certain constraints on the input, such as the requirement that
>identifiers be declared before they are used, cannot be described by a
>context-free grammar."
>
>
>If there are places where the grammar needs to be tightened up, let me
>know.
>
>--
>Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
>
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip



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In section 16.1 of RFC2543bis there is an example of Registration. I
have
a couple of questions about this scenario. I would appreciate any
responses.

1) What would be the To header of the REGISTER that T. Watson sends to
the registrar at example.com. Would it be tawatson@example.com or
tawatson@bell-tel.com ?? As per the description in section 4.2.6, To
header
domain could be different from the domain of the Request-URI in the
REGISTER.

2) Let us say x@bell-tel.com (From: x@bell-tel.com) sends an INVITE to
the
proxy at bell-tel.com and the Request-URI is INVITE watson@bell-tel.com.
As per the latest registration, the Contact is tawatson@example.com, so 
the proxy at bell-tel.com rewrites the Request-URI and sends it to the
proxy at example.com - INVITE tawatson@example.com. Now what is the
search
key at the registrar - is it user@domain or simply user. If it is simply
user,
then domain of To header in REGISTER is meaningless. If it is
user@domain,
then To header of the REGISTER sent to the registrar at example.com must
be
tawatson@example.com.

3) What if there already exists a user tawatson at the domain
example.com.
Should the registrar at example.com reject the registration ?? If yes,
what response code should be used ??

Thanks,
Shail


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You definitely want to handle things like this. There are many
innovative services that are enabled by placing URLs like this in web
pages, Contact headers in redirects, or inside of instant messages.

-Jonathan R.

Rohan Mahy wrote:
> 
> Hi,
> 
> I just want to make sure that we can handle/describe the escaping of a
> nested monster such as this:  (sorrry for the wrapping at hyphens)
> 
> <a
> href="<sip:userA.butler@sip.cisco.com:5060;method=TRANSFER?Transfer-To=<sip:
> +14083079999@sip.cisco.com:5060;user=phone?Request-Disposition=no-fork&Accep
> t-Contact=*;mobility="mobile",*;media="audio">&Request-Disposition=parallel>">
> 
> this would be a SIP URI inside an HTML page, that asks userA's "Butler"
> (wherever he/she might be) to call (presumably userA's) cell phone.
> 
> this particular example may be a bit contrived, but i can come up with
> legitimate uses of this kind of functionality.
> 
> thanks,
> -rohan
> 
> At 11:37 AM 4/29/00 , Henning Schulzrinne wrote:
> >Maybe time to summarize and see if there's an issue left:
> >
> >- The host part of the URL cannot contain %, either as itself or as an
> >escape mechanism:
> >
> >host          = hostname | IPv4address
> >hostname      = *( domainlabel "." ) toplabel [ "." ]
> >domainlabel   = alphanum | alphanum *( alphanum | "-" ) alphanum
> >toplabel      = alpha | alpha *( alphanum | "-" ) alphanum
> >alphanum      = alpha | digit
> >
> >- For "token" elements in URLs, this is a bit tricky, since we have two
> >escape mechanisms that are intersecting:
> >
> >   (1) the normal token definition, i.e., any letter, digit, and a large
> >number of symbols (-, #, $, &, %, ^, *, _, +, |, ~, ')
> >
> >   (2) the rules for URLs, where some of these characters (&, +) are
> >reserved and thus need to be escaped if there's the possibility of
> >confusion with a separator. As far as I can tell, neither of them are
> >syntactically relevant for SIP URLs in this context (the & is only used
> >to separate 'header' items), so neither of them would need to be
> >escaped.
> >
> >Thus, it might be better to define
> >
> >other-param = pname [ "=" pvalue ]
> >
> >pname     = 1*tokenchar
> >pvalue    = 1*tokenchar | quoted-string
> >tokenchar = alphanum | escaped | "!" | "'" | "*" | "."
> >             | "^" | "_" | "~" | "|" | "%" | "-" | "#" | "/" | ":" | "@"
> >| "&" | "+" | "$"
> >
> >(since "," is not valid in token, I didn't include it. Any that I'm
> >missing?)
> >
> >Here, escaped can only refer to the legal characters (alphanum and the
> >others) to maintain "tokenness", but I can't express this in the BNF.
> >However, in practice this is not a problem except for spec-writers since
> >the pname and pvalue tokens have to be drawn from an enumeration, so
> >that most character combinations are invalid simply because they are not
> >valid parameter names.
> >
> >- To make matters slightly more confusing, we do have a different
> >potential problem, namely the " in the quoted string. The <"> is a
> >"delims" character and thus must be escaped (RFC 2396, pg. 9/10). One
> >could argue that it only needs to be escaped if there's a possibility
> >for confusion, e.g., on a web page or if the URL appears in a quoted
> >string. However, either should be accepted by the parser.
> >
> >
> >- In URLs, %hex hex (where allowed) is always interpreted as an escape
> >mechanism. It is mechanically translated to the equivalent character,
> >while maintaining its distinct identity as a non-separator (if it could
> >be treated as such). Thus,
> >
> >foo%40bar@host.com
> >
> >can't just be translated into foo@bar@host.com and then be parsed, since
> >the result would not be as expected. This is not fundamentally different
> >from other escape mechanisms for delimiters, like \" or "".
> >
> >Thus, one should *first* split the URL into components and then
> >translate those parts where escaping is possible (which is pretty much
> >anywhere except in scheme and host).
> >
> >- As was pointed out, BNF specifies syntax. It cannot express certain
> >things, including
> >   . restrictions on ordering  (an | does not imply that the elements
> >have to appear in order)
> >   . any restriction on duplication
> >   . the role of identifiers or restrictions on their names (e.g., "int
> >int" may be legal in a BNF specification but the compiler may still
> >complain)
> >
> >   For details, see for example, pg. 172 of Aho/Sethi/Ullman:
> >
> >   "Certain constraints on the input, such as the requirement that
> >identifiers be declared before they are used, cannot be described by a
> >context-free grammar."
> >
> >
> >If there are places where the grammar needs to be tightened up, let me
> >know.
> >
> >--
> >Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> >
> >
> >_______________________________________________
> >SIP mailing list
> >SIP@lists.bell-labs.com
> >http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Shail Bhatnagar wrote:
> 
> In section 16.1 of RFC2543bis there is an example of Registration. I
> have
> a couple of questions about this scenario. I would appreciate any
> responses.
> 
> 1) What would be the To header of the REGISTER that T. Watson sends to
> the registrar at example.com. Would it be tawatson@example.com or
> tawatson@bell-tel.com ?? As per the description in section 4.2.6, To
> header
> domain could be different from the domain of the Request-URI in the
> REGISTER.

Most likely tawatson@example.com. This is because if some UAC or proxy
wants to send a request to tawatson@example.com, it will use DNS to find
the SIP server for example.com, and forward the request there. As a
result, watson will want to register tawatson@example.com with the
server for which requests to example.com will be routed. The spec does
not prohibit registering tawatson@bell-tel.com with example.com.
However, example.com is not likely to receive requests with domains of
bell-tel.com in the request URI.

> 
> 2) Let us say x@bell-tel.com (From: x@bell-tel.com) sends an INVITE to
> the
> proxy at bell-tel.com and the Request-URI is INVITE watson@bell-tel.com.
> As per the latest registration, the Contact is tawatson@example.com, so
> the proxy at bell-tel.com rewrites the Request-URI and sends it to the
> proxy at example.com - INVITE tawatson@example.com. Now what is the
> search
> key at the registrar - is it user@domain or simply user. If it is simply
> user,
> then domain of To header in REGISTER is meaningless. If it is
> user@domain,
> then To header of the REGISTER sent to the registrar at example.com must
> be
> tawatson@example.com.

Thats your choice. If your proxy only handles a single domain, using
user alone as a lookup key into your database is fine. If your proxy
handles multiple domains, you will want to use the complete user@host.


> 
> 3) What if there already exists a user tawatson at the domain
> example.com.
> Should the registrar at example.com reject the registration ?? If yes,
> what response code should be used ??

No, the new Contact would be added to the existing registration. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From: Giuseppe Ricagni <giuseppe.ricagni@italtel.it>
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To: James Yu <james.yu@neustar.com>
Cc: "'James Kempf'" <James.Kempf@Eng.Sun.COM>, sip@lists.bell-labs.com
Subject: Re: [SIP] SIP Info Method and IS-41 Cellular Signalling Protocol
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James Yu wrote:

> James,
>
> Your question can be addressed by sigtran - SS7 transport over IP.  There
> are a few adaptation layers to choose including the newly proposed SUA (for
> SCCP users such as IS-41).
>
> James Yu
>

Yes.

You definetly don' t need SIP if you only want to tunnel legacy
signalling
protocols over IP: SIGTRAN does the job much better (providing for
endpoint
redundancy, scalability, through the concept of multiple streams within
an
association, encompassing a very efficient SACKs implementation that
gives its
best in case of retranmission of small chunks of information, just like
signaling messages, and.... really much more).

IMHO, you should tunnel in SIP (over SCTP, maybe) when you really add
something
to the base protocol, e.g by means of the SDP attributes SIP can carry,
that
allow for the negotiation of much richer sessions.

The point is that this is actually the case, at least in many 3G
cellular
scenarios.

A completely new (SIP-based if SIP can provide all the necessary
features)
approach  could be also welcome in many cases.

Giuseppe



----------------------------------------------------------
Giuseppe RICAGNI
System Architecture and Product Planning
Broadband Networks
Italtel
via Reiss Romoli - C10
20019 Castelletto di Settimo Milanese (MILANO)
ITALY

mailto:giuseppe.ricagni@italtel.it
----------------------------------------------------------




>
> > -----Original Message-----
> > From: James Kempf [mailto:James.Kempf@Eng.Sun.COM]
> > Sent: Friday, April 14, 2000 5:59 PM
> > To: sip@lists.bell-labs.com
> > Cc: James.Kempf@Eng.Sun.COM
> > Subject: [SIP] SIP Info Method and IS-41 Cellular Signalling Protocol
> >
> >
> > I'm working on a design for replacing the IS-41 North
> > American cellular
> > signalling protocol with IP protocols. I had a couple of
> > questions about
> > how the SIP INFO method, or potentially another SIP method, might
> > be used.
> >
> > 1) In the introduction to draft-ietf-sip-info-method-03.txt, exchange
> > of radio signal strength information is explicitly mentioned as
> > an application of the SIP INFO method. The HandoffMeasurementRequest
> > and HandoffMeasurementRequest2 messages in IS-41 are sent by
> > the anchor
> > Mobile Switching Center (MSC) to a potential serving MSC to
> > obtain information
> > about the signal strength of a Mobile Station (MS) in order
> > to determine
> > if the potential serving MSC is a better candidate for handling the
> > mobile's call (called "hard handoff" in cellular jargon).
> > They could be
> > replaced with a SIP INFO method. There are three problems:
> >
> >       a) The radio information is always solicited by the anchor MSC,
> >       and from the ID, the SIP INFO method seems like it is
> >       sent unsolicted. I suppose one could come up with two INFO
> >       methods, one to solicit the radio information and one to
> >       return it, but this seems like a hack somehow.
> >
> >       b) In the ID, it explicitly states that the INFO method is
> >       only applicable when there is a call in progress. Problem is,
> >       there isn't a call in progress between the anchor MSC and
> >       the serving MSC, that's why the anchor MSC is soliciting
> >       the radio quality information in the first place.
> >
> >       c) The INFO method seems like it is meant to be sent between
> >       SIP UAC's, while the HandoffMeasurementRequest(2) methods
> >       are primarily designed to be sent between network elements.
> >       A SIP replacement would probably be sent between the
> > last hop SIP
> >       proxies acting as Call Agents (CAs) for the call, which replaces
> >       the MSC in an all IP network.
> >
> > 2) There are a bunch of IS-41 messages that involve transmitting
> > feature or other information between the MS's Home Location
> > Register (HLR) and the MS. These are initiated by either the serving
> > MSC or the HLR. Examples are InformationDirective, which is used
> > by the HLR to provide notifications to an idle MS, and FeatureRequest,
> > which sends particular features to a potentially idle MS (features
> > being things like messages or playing particular tones). Again, these
> > seem like candidates to replace with the SIP INFO message, but
> > the potential lack of any call in progress would prohibit that.
> >
> > The alternative I could see would be establishing a session in order
> > to then send the information via the INFO method, but that seems like
> > overkill. And in the case of HandoffMeasurementRequest(2), the
> > message isn't intended for a SIP UAC anyway.
> >
> > Maybe SIP is the wrong protocol to consider, since it is dedicated
> > to setting up and maintaining sessions, and not to exchanging
> > information
> > between network elements or between network elements and terminals?
> >
> > Can anybody provide some guidance?
> >
> >               jak
> >
> >
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> >
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--
----------------------------------------------------------
Giuseppe RICAGNI
System Architecture and Product Planning
Broadband Networks
Italtel
via Reiss Romoli - C10
20019 Castelletto di Settimo Milanese (MILANO)
ITALY

mailto:giuseppe.ricagni@italtel.it
----------------------------------------------------------



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Cc: sip@lists.bell-labs.com
Subject: RE: [SIP] SIP Info Method and IS-41 Cellular Signalling Protocol
Date: Tue, 2 May 2000 07:50:34 -0500 
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Cellular intersystem operations can be separated into two types: transaction
based (e.g., for roaming support and handoff measurement) and call- or
service-related (for call setup during handoff and for intelligent network
call/service control).  Call-related operations certainly can be supported
by SIP.  Service control related operations probably are best supported by
MGCP/megaco if WIN/CAP are not to be used.  If WIN/CAP are used for
call/service control, sigtran would be the choice to carry the protocol.
Transaction based operations probably can just use sigtran to transport the
messages.  You can always add new parameters to IS-41 or GSM MAP to carry
SDP information. 

My earlier comment was about the handoff measurement part where no call set
up is needed. But I think you are looking at SIP or something else to
totally replace IS-41 (so as not to enhance IS-41 to support 3G)- so please
continue the search/investigation.

James  


> -----Original Message-----
> From: Giuseppe Ricagni [mailto:giuseppe.ricagni@italtel.it]
> Sent: Tuesday, May 02, 2000 6:39 AM
> To: James Yu
> Cc: 'James Kempf'; sip@lists.bell-labs.com
> Subject: Re: [SIP] SIP Info Method and IS-41 Cellular Signalling
> Protocol
> 
> 
> James Yu wrote:
> 
> > James,
> >
> > Your question can be addressed by sigtran - SS7 transport 
> over IP.  There
> > are a few adaptation layers to choose including the newly 
> proposed SUA (for
> > SCCP users such as IS-41).
> >
> > James Yu
> >
> 
> Yes.
> 
> You definetly don' t need SIP if you only want to tunnel legacy
> signalling
> protocols over IP: SIGTRAN does the job much better (providing for
> endpoint
> redundancy, scalability, through the concept of multiple 
> streams within
> an
> association, encompassing a very efficient SACKs implementation that
> gives its
> best in case of retranmission of small chunks of information, 
> just like
> signaling messages, and.... really much more).
> 
> IMHO, you should tunnel in SIP (over SCTP, maybe) when you really add
> something
> to the base protocol, e.g by means of the SDP attributes SIP 
> can carry,
> that
> allow for the negotiation of much richer sessions.
> 
> The point is that this is actually the case, at least in many 3G
> cellular
> scenarios.
> 
> A completely new (SIP-based if SIP can provide all the necessary
> features)
> approach  could be also welcome in many cases.
> 
> Giuseppe
> 
> 
> 
> ----------------------------------------------------------
> Giuseppe RICAGNI
> System Architecture and Product Planning
> Broadband Networks
> Italtel
> via Reiss Romoli - C10
> 20019 Castelletto di Settimo Milanese (MILANO)
> ITALY
> 
> mailto:giuseppe.ricagni@italtel.it
> ----------------------------------------------------------
> 


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Date: Tue, 02 May 2000 13:18:24 -0400
From: Shail Bhatnagar <shbhatna@cisco.com>
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: SIP IETF <sip@lists.research.bell-labs.com>
Subject: Re: [SIP] Second try : Questions about REGISTER
References: <390DB967.CBA8060D@cisco.com> <390E603C.10ED09F5@dynamicsoft.com>
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Jonathan, Appreciate your responses. See embedded .
Shail

Jonathan Rosenberg wrote:
> 
> Shail Bhatnagar wrote:
> >
> > In section 16.1 of RFC2543bis there is an example of Registration. I
> > have
> > a couple of questions about this scenario. I would appreciate any
> > responses.
> >
> > 1) What would be the To header of the REGISTER that T. Watson sends to
> > the registrar at example.com. Would it be tawatson@example.com or
> > tawatson@bell-tel.com ?? As per the description in section 4.2.6, To
> > header
> > domain could be different from the domain of the Request-URI in the
> > REGISTER.
> 
> Most likely tawatson@example.com. This is because if some UAC or proxy
> wants to send a request to tawatson@example.com, it will use DNS to find
> the SIP server for example.com, and forward the request there. As a
> result, watson will want to register tawatson@example.com with the
> server for which requests to example.com will be routed. The spec does
> not prohibit registering tawatson@bell-tel.com with example.com.
> However, example.com is not likely to receive requests with domains of
> bell-tel.com in the request URI.

This is exactly my point of confusion. The text in 4.2.6 is actually
raising
more questions than answering. I cannot think of a case when a mismatch
in
the domain of To header and Request-URI would be useful. If the proxy
services
more than 2 domains, the domain in To header and Request-URI would still
be
same or may be I am missing something.


> 
> >
> > 2) Let us say x@bell-tel.com (From: x@bell-tel.com) sends an INVITE to
> > the
> > proxy at bell-tel.com and the Request-URI is INVITE watson@bell-tel.com.
> > As per the latest registration, the Contact is tawatson@example.com, so
> > the proxy at bell-tel.com rewrites the Request-URI and sends it to the
> > proxy at example.com - INVITE tawatson@example.com. Now what is the
> > search
> > key at the registrar - is it user@domain or simply user. If it is simply
> > user,
> > then domain of To header in REGISTER is meaningless. If it is
> > user@domain,
> > then To header of the REGISTER sent to the registrar at example.com must
> > be
> > tawatson@example.com.
> 
> Thats your choice. If your proxy only handles a single domain, using
> user alone as a lookup key into your database is fine. If your proxy
> handles multiple domains, you will want to use the complete user@host.
> 
> >
> > 3) What if there already exists a user tawatson at the domain
> > example.com.
> > Should the registrar at example.com reject the registration ?? If yes,
> > what response code should be used ??
> 
> No, the new Contact would be added to the existing registration.


> 
> -Jonathan R.
> 
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
>


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I have a question regarding Jonathan's answer to the Shail's third
question.
Do we assume that tawatson has an account created for himself at
example.com or he is just a visitor at example.com?
I assume the first.
What would be the behaviour is tawatson@bell-tell.com would be just a
visitor at example.com?

=20
-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Monday, May 01, 2000 9:58 PM
To: Shail Bhatnagar
Cc: SIP IETF
Subject: Re: [SIP] Second try : Questions about REGISTER




Shail Bhatnagar wrote:
>=20
> In section 16.1 of RFC2543bis there is an example of Registration. I
> have
> a couple of questions about this scenario. I would appreciate any
> responses.
>=20
> 1) What would be the To header of the REGISTER that T. Watson sends to
> the registrar at example.com. Would it be tawatson@example.com or
> tawatson@bell-tel.com ?? As per the description in section 4.2.6, To
> header
> domain could be different from the domain of the Request-URI in the
> REGISTER.

Most likely tawatson@example.com. This is because if some UAC or proxy
wants to send a request to tawatson@example.com, it will use DNS to find
the SIP server for example.com, and forward the request there. As a
result, watson will want to register tawatson@example.com with the
server for which requests to example.com will be routed. The spec does
not prohibit registering tawatson@bell-tel.com with example.com.
However, example.com is not likely to receive requests with domains of
bell-tel.com in the request URI.

>=20
> 2) Let us say x@bell-tel.com (From: x@bell-tel.com) sends an INVITE to
> the
> proxy at bell-tel.com and the Request-URI is INVITE
watson@bell-tel.com.
> As per the latest registration, the Contact is tawatson@example.com,
so
> the proxy at bell-tel.com rewrites the Request-URI and sends it to the
> proxy at example.com - INVITE tawatson@example.com. Now what is the
> search
> key at the registrar - is it user@domain or simply user. If it is
simply
> user,
> then domain of To header in REGISTER is meaningless. If it is
> user@domain,
> then To header of the REGISTER sent to the registrar at example.com
must
> be
> tawatson@example.com.

Thats your choice. If your proxy only handles a single domain, using
user alone as a lookup key into your database is fine. If your proxy
handles multiple domains, you will want to use the complete user@host.


>=20
> 3) What if there already exists a user tawatson at the domain
> example.com.
> Should the registrar at example.com reject the registration ?? If yes,
> what response code should be used ??

No, the new Contact would be added to the existing registration.=20

-Jonathan R.

--=20
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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<BR>

<P><FONT SIZE=3D2>I have a question regarding Jonathan's answer to the =
Shail's third question.</FONT>

<BR><FONT SIZE=3D2>Do we assume that tawatson has an account created for =
himself at example.com or he is just a visitor at example.com?</FONT>

<BR><FONT SIZE=3D2>I assume the first.</FONT>

<BR><FONT SIZE=3D2>What would be the behaviour is tawatson@bell-tell.com =
would be just a visitor at example.com?</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;</FONT>

<BR><FONT SIZE=3D2>-----Original Message-----</FONT>

<BR><FONT SIZE=3D2>From: Jonathan Rosenberg [<A =
HREF=3D"mailto:jdrosen@dynamicsoft.com">mailto:jdrosen@dynamicsoft.com</A=
>]</FONT>

<BR><FONT SIZE=3D2>Sent: Monday, May 01, 2000 9:58 PM</FONT>

<BR><FONT SIZE=3D2>To: Shail Bhatnagar</FONT>

<BR><FONT SIZE=3D2>Cc: SIP IETF</FONT>

<BR><FONT SIZE=3D2>Subject: Re: [SIP] Second try : Questions about =
REGISTER</FONT>
</P>
<BR>
<BR>
<BR>

<P><FONT SIZE=3D2>Shail Bhatnagar wrote:</FONT>

<BR><FONT SIZE=3D2>&gt; </FONT>

<BR><FONT SIZE=3D2>&gt; In section 16.1 of RFC2543bis there is an =
example of Registration. I</FONT>

<BR><FONT SIZE=3D2>&gt; have</FONT>

<BR><FONT SIZE=3D2>&gt; a couple of questions about this scenario. I =
would appreciate any</FONT>

<BR><FONT SIZE=3D2>&gt; responses.</FONT>

<BR><FONT SIZE=3D2>&gt; </FONT>

<BR><FONT SIZE=3D2>&gt; 1) What would be the To header of the REGISTER =
that T. Watson sends to</FONT>

<BR><FONT SIZE=3D2>&gt; the registrar at example.com. Would it be =
tawatson@example.com or</FONT>

<BR><FONT SIZE=3D2>&gt; tawatson@bell-tel.com ?? As per the description =
in section 4.2.6, To</FONT>

<BR><FONT SIZE=3D2>&gt; header</FONT>

<BR><FONT SIZE=3D2>&gt; domain could be different from the domain of the =
Request-URI in the</FONT>

<BR><FONT SIZE=3D2>&gt; REGISTER.</FONT>
</P>

<P><FONT SIZE=3D2>Most likely tawatson@example.com. This is because if =
some UAC or proxy</FONT>

<BR><FONT SIZE=3D2>wants to send a request to tawatson@example.com, it =
will use DNS to find</FONT>

<BR><FONT SIZE=3D2>the SIP server for example.com, and forward the =
request there. As a</FONT>

<BR><FONT SIZE=3D2>result, watson will want to register =
tawatson@example.com with the</FONT>

<BR><FONT SIZE=3D2>server for which requests to example.com will be =
routed. The spec does</FONT>

<BR><FONT SIZE=3D2>not prohibit registering tawatson@bell-tel.com with =
example.com.</FONT>

<BR><FONT SIZE=3D2>However, example.com is not likely to receive =
requests with domains of</FONT>

<BR><FONT SIZE=3D2>bell-tel.com in the request URI.</FONT>
</P>

<P><FONT SIZE=3D2>&gt; </FONT>

<BR><FONT SIZE=3D2>&gt; 2) Let us say x@bell-tel.com (From: =
x@bell-tel.com) sends an INVITE to</FONT>

<BR><FONT SIZE=3D2>&gt; the</FONT>

<BR><FONT SIZE=3D2>&gt; proxy at bell-tel.com and the Request-URI is =
INVITE watson@bell-tel.com.</FONT>

<BR><FONT SIZE=3D2>&gt; As per the latest registration, the Contact is =
tawatson@example.com, so</FONT>

<BR><FONT SIZE=3D2>&gt; the proxy at bell-tel.com rewrites the =
Request-URI and sends it to the</FONT>

<BR><FONT SIZE=3D2>&gt; proxy at example.com - INVITE =
tawatson@example.com. Now what is the</FONT>

<BR><FONT SIZE=3D2>&gt; search</FONT>

<BR><FONT SIZE=3D2>&gt; key at the registrar - is it user@domain or =
simply user. If it is simply</FONT>

<BR><FONT SIZE=3D2>&gt; user,</FONT>

<BR><FONT SIZE=3D2>&gt; then domain of To header in REGISTER is =
meaningless. If it is</FONT>

<BR><FONT SIZE=3D2>&gt; user@domain,</FONT>

<BR><FONT SIZE=3D2>&gt; then To header of the REGISTER sent to the =
registrar at example.com must</FONT>

<BR><FONT SIZE=3D2>&gt; be</FONT>

<BR><FONT SIZE=3D2>&gt; tawatson@example.com.</FONT>
</P>

<P><FONT SIZE=3D2>Thats your choice. If your proxy only handles a single =
domain, using</FONT>

<BR><FONT SIZE=3D2>user alone as a lookup key into your database is =
fine. If your proxy</FONT>

<BR><FONT SIZE=3D2>handles multiple domains, you will want to use the =
complete user@host.</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt; </FONT>

<BR><FONT SIZE=3D2>&gt; 3) What if there already exists a user tawatson =
at the domain</FONT>

<BR><FONT SIZE=3D2>&gt; example.com.</FONT>

<BR><FONT SIZE=3D2>&gt; Should the registrar at example.com reject the =
registration ?? If yes,</FONT>

<BR><FONT SIZE=3D2>&gt; what response code should be used ??</FONT>
</P>

<P><FONT SIZE=3D2>No, the new Contact would be added to the existing =
registration. </FONT>
</P>

<P><FONT SIZE=3D2>-Jonathan R.</FONT>
</P>

<P><FONT SIZE=3D2>-- </FONT>

<BR><FONT SIZE=3D2>Jonathan D. =
Rosenberg&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 72 =
Eagle Rock Ave.</FONT>

<BR><FONT SIZE=3D2>Chief =
Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; First Floor</FONT>

<BR><FONT =
SIZE=3D2>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>jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp; FAX:&nbsp;&nbsp; (732) 741-4778</FONT>

<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.cs.columbia.edu/~jdrosen">http://www.cs.columbia.edu/~=
jdrosen</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: (732) =
741-7244</FONT>

<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.dynamicsoft.com">http://www.dynamicsoft.com</A></FONT>=

</P>
<BR>

<P><FONT SIZE=3D2>_______________________________________________</FONT>

<BR><FONT SIZE=3D2>SIP mailing list</FONT>

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Date: Tue, 02 May 2000 17:57:19 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Shail Bhatnagar <shbhatna@cisco.com>
Cc: SIP IETF <sip@lists.research.bell-labs.com>
Subject: Re: [SIP] Second try : Questions about REGISTER
References: <390DB967.CBA8060D@cisco.com> <390E603C.10ED09F5@dynamicsoft.com> <390F0DE0.EED96548@cisco.com>
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Shail Bhatnagar wrote:
> 
> > > 1) What would be the To header of the REGISTER that T. Watson sends to
> > > the registrar at example.com. Would it be tawatson@example.com or
> > > tawatson@bell-tel.com ?? As per the description in section 4.2.6, To
> > > header
> > > domain could be different from the domain of the Request-URI in the
> > > REGISTER.
> >
> > Most likely tawatson@example.com. This is because if some UAC or proxy
> > wants to send a request to tawatson@example.com, it will use DNS to find
> > the SIP server for example.com, and forward the request there. As a
> > result, watson will want to register tawatson@example.com with the
> > server for which requests to example.com will be routed. The spec does
> > not prohibit registering tawatson@bell-tel.com with example.com.
> > However, example.com is not likely to receive requests with domains of
> > bell-tel.com in the request URI.
> 
> This is exactly my point of confusion. The text in 4.2.6 is actually
> raising
> more questions than answering. I cannot think of a case when a mismatch
> in
> the domain of To header and Request-URI would be useful. If the proxy
> services
> more than 2 domains, the domain in To header and Request-URI would still
> be
> same or may be I am missing something.

I think this text just requires additional clarification. There are
cases where this can work (namely, when there is specialized routing
logic at a server), but we need to give an example and the limitations.
For 99% of the applications, the domain in the request URI and To field
will be identical.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Tue May  2 18:20:07 2000
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From: Jonathan Lennox <lennox@cs.columbia.edu>
To: sip@lists.research.bell-labs.com
Subject: [SIP] Problem with the syntax of URI extension parameters
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RFC 2543 (and the current bis draft) defines the BNF for SIP URLs' extension
parameters as follows:

  other-param     = ( token | ( token "=" ( token | quoted-string )))

The problem with this is that RFC 2396 (URI Generic Syntax) specifically
excludes quote marks from URIs, since they're a common URI delimeter in text
documents and protocol fields.  What's more, both the "token" and
"quoted-string" syntax elements come from a very different part of the SIP
BNF; and no other URI syntax, as far as I know, uses anything like them.

I think it'd be much more in the spirit of the generic URI syntax to have 
other-param be defined like this:

  other-param     = pname | pname "=" pvalue
  pname           = 1*paramchar
  pvalue          = 1*paramchar
  paramchar       = param-reserved | unreserved | escaped
  param-reserved  = "/" | "?" | ":" | "@" | "&" | "+" | "$" | ","

That is, special characters get %-escaped rather than quoted.
"param-reserved" above is the same as the general "reserved" with ";" and
"=", the parameter delimiters, omitted.

This would, however, be an incompatible change to the SIP spec.  What would
people (especially the spec authors) think about this?

-- 
Jonathan Lennox
lennox@cs.columbia.edu


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Date: Tue, 02 May 2000 17:24:49 -0500
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: [SIP] Second try : Questions about REGISTER
In-reply-to: <390F4F3F.62D59146@dynamicsoft.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Shail Bhatnagar <shbhatna@cisco.com>
Cc: SIP IETF <sip@lists.research.bell-labs.com>
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> > This is exactly my point of confusion. The text in 4.2.6 is actually
> > raising
> > more questions than answering. I cannot think of a case when a mismatch
> > in
> > the domain of To header and Request-URI would be useful. If the proxy
> > services
> > more than 2 domains, the domain in To header and Request-URI would still
> > be
> > same or may be I am missing something.
>
> I think this text just requires additional clarification. There are
> cases where this can work (namely, when there is specialized routing
> logic at a server), but we need to give an example and the limitations.
> For 99% of the applications, the domain in the request URI and To field
> will be identical.

True, it is unlikley that an original UAC will generate a request with
variant fields.

However, I might suggest that a mismatch between request-URI and To: field
will inevitably occur if a previous proxy in the path has applied routing
logic that results in a new request-URI being applied.

For example, let's say my work proxy forwards to my home proxy:

So an original INVITE might contain:

INVITE sip:dean.willis@wcom.com SIP/2.0
To: Dean Willis <sip:dean.willis@wcom.com>

and after leaving the wcom.com proxy (which, based on time-of-day, forwards
to my house)

INVITE sip:dwillis@sip.softarmor.com SIP/2.0
TO: Dean Willis <sip:dean.willis@wcom.com>

So the destination UAS (my trusty sip-phone) would see an INVITE in which
the request-URI does not match the To: field.

--
Dean



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Date: Tue, 02 May 2000 18:53:12 -0400
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To: Dean Willis <dean.willis@wcom.com>
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        SIP IETF <sip@lists.research.bell-labs.com>
Subject: Re: [SIP] Second try : Questions about REGISTER
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I was referring to REGISTER request, not to other requests.


Dean Willis wrote:
> 
> > > This is exactly my point of confusion. The text in 4.2.6 is actually
> > > raising
> > > more questions than answering. I cannot think of a case when a mismatch
> > > in
> > > the domain of To header and Request-URI would be useful. If the proxy
> > > services
> > > more than 2 domains, the domain in To header and Request-URI would still
> > > be
> > > same or may be I am missing something.
> >
> > I think this text just requires additional clarification. There are
> > cases where this can work (namely, when there is specialized routing
> > logic at a server), but we need to give an example and the limitations.
> > For 99% of the applications, the domain in the request URI and To field
> > will be identical.
> 
> True, it is unlikley that an original UAC will generate a request with
> variant fields.
> 
> However, I might suggest that a mismatch between request-URI and To: field
> will inevitably occur if a previous proxy in the path has applied routing
> logic that results in a new request-URI being applied.
> 
> For example, let's say my work proxy forwards to my home proxy:
> 
> So an original INVITE might contain:
> 
> INVITE sip:dean.willis@wcom.com SIP/2.0
> To: Dean Willis <sip:dean.willis@wcom.com>
> 
> and after leaving the wcom.com proxy (which, based on time-of-day, forwards
> to my house)
> 
> INVITE sip:dwillis@sip.softarmor.com SIP/2.0
> TO: Dean Willis <sip:dean.willis@wcom.com>
> 
> So the destination UAS (my trusty sip-phone) would see an INVITE in which
> the request-URI does not match the To: field.
> 
> --
> Dean


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Date: Tue, 02 May 2000 18:22:29 -0500
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: [SIP] Second try : Questions about REGISTER
In-reply-to: <390F5C58.AFEE47C1@cisco.com>
To: shbhatna@cisco.com
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
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Pardon me for not carrying the thought out. I committed premature sending
again. Need keyboard viagra or something.

Ok, we've seen from my previous example how an INVITE with a To: that does
not match the request-URI can show up at my home proxy.

What does my home proxy DO with that INVITE?

It could follow the usual wisdom, and simply process the INVITE according to
the rules applicable to the entity specified by the request-URI.

What if I wanted a little more power in my call handling? For example, I
might want my home proxy to divert calls that are To:
dwillis@sip.softarmor.com to be diverted to my cell phone, and calls that
are To: dean.willis@wcom.com (calls forwarded from work) to ring my desk
phone, then be diverted to my unified messaging box if there is no answer.

This implies logic processing in the home proxy based on the value of the
To: header. How do I provision this logic?

One possibility would be to put it in the proxy's config file as a script or
action table.

What if I wanted to use dynamic registration, perhaps with a CPL upload, to
change proxy behavior?

How would I specify that one register affects a different logical target or
"To:" than another?

Two possibilities:

1) There is only one registration, and the singular uploaded CPL includes
To: differentiation logic.

2) I use a register message which indicates second logical destination in
the To: field, and have some other way of indicating a request-chaining
logic.

As Jonathan says, 99%+ of all registers will have a request-URI that matches
a To:. I think, however, there are services that have not been completely
defined which would take advantage of REGISTER messages which have different
request-URI and To: fields.

Personally, I wonder why we didn't just go ahead and include the user part
in the request-URI. This would allow my request-chaining REGISTER to be
fairly explicitly stated and solve lots of messiness.

--
Dean

> -----Original Message-----
> From: shbhatna@cisco.com [mailto:shbhatna@cisco.com]
> Sent: Tuesday, May 02, 2000 5:53 PM
> To: Dean Willis
> Cc: Jonathan Rosenberg; SIP IETF
> Subject: Re: [SIP] Second try : Questions about REGISTER
>
>
> I was referring to REGISTER request, not to other requests.
>
>
> Dean Willis wrote:
> >
> > > > This is exactly my point of confusion. The text in 4.2.6 is actually
> > > > raising
> > > > more questions than answering. I cannot think of a case
> when a mismatch
> > > > in
> > > > the domain of To header and Request-URI would be useful. If
> the proxy
> > > > services
> > > > more than 2 domains, the domain in To header and
> Request-URI would still
> > > > be
> > > > same or may be I am missing something.
> > >
> > > I think this text just requires additional clarification. There are
> > > cases where this can work (namely, when there is specialized routing
> > > logic at a server), but we need to give an example and the
> limitations.
> > > For 99% of the applications, the domain in the request URI
> and To field
> > > will be identical.
> >
> > True, it is unlikley that an original UAC will generate a request with
> > variant fields.
> >
> > However, I might suggest that a mismatch between request-URI
> and To: field
> > will inevitably occur if a previous proxy in the path has
> applied routing
> > logic that results in a new request-URI being applied.
> >
> > For example, let's say my work proxy forwards to my home proxy:
> >
> > So an original INVITE might contain:
> >
> > INVITE sip:dean.willis@wcom.com SIP/2.0
> > To: Dean Willis <sip:dean.willis@wcom.com>
> >
> > and after leaving the wcom.com proxy (which, based on
> time-of-day, forwards
> > to my house)
> >
> > INVITE sip:dwillis@sip.softarmor.com SIP/2.0
> > TO: Dean Willis <sip:dean.willis@wcom.com>
> >
> > So the destination UAS (my trusty sip-phone) would see an
> INVITE in which
> > the request-URI does not match the To: field.
> >
> > --
> > Dean
>



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To: jdrosen@dynamicsoft.com, shbhatna@cisco.com
Subject: Re: [SIP] Second try : Questions about REGISTER
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> This is exactly my point of confusion. The text in 4.2.6 is actually
> raising
> more questions than answering. I cannot think of a case when a mismatch
> in
> the domain of To header and Request-URI would be useful. If the proxy
> services
> more than 2 domains, the domain in To header and Request-URI would still
> be
> same or may be I am missing something.
> 

Virtual hosting. A single SIP proxy accepts (REGISTER) requests for multiple domains.
Depending on the domain, the proxy forwards the REGISTER request to a particular
provisioned registrar, in the process re-writing the domain of the request URI according
to that registrar's needs/whims. This could even involve forking of the REGISTER request.
The To: remains (of course) unchanged.

Or maybe load balancing. A single domain is split into logical sub-domains based on the
a combination of the user/To/From/whatever by a front-end proxy. The request-URI is
rewritten to reflect the particular sub-domain of the registrar to which the REGISTER 
request ultimately gets forwarded. The To: remains (of course) unchanged.

Or as indicated on page 28 under the Request-URI description, the registrar acting
as a VLR accepts registrations for domains other than the domain given in the 
Request-URI.

The To: header determines the registration that is affected. The request-URI 
determines the domain of the registrar. These are logically separate concepts and there
is no good reason to mandate that the domain in the To: is the same as the domain
in the Request-URI

Sean Olson <sean.olson@ericsson.com>


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> Personally, I wonder why we didn't just go ahead and include the user part
> in the request-URI. This would allow my request-chaining REGISTER to be
> fairly explicitly stated and solve lots of messiness.
> 
> --
> Dean
> 

I second that :) I've never understood the rationale behind excluding the user part in
the Request-URI of a REGISTER request.

--
Sean Olson <sean.olson@ericsson.com>


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To: sip@lists.research.bell-labs.com, lennox@cs.columbia.edu
Subject: Re: [SIP] Problem with the syntax of URI extension parameters
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> RFC 2543 (and the current bis draft) defines the BNF for SIP URLs' extension
> parameters as follows:
> 
>   other-param     = ( token | ( token "=" ( token | quoted-string )))
> 
> The problem with this is that RFC 2396 (URI Generic Syntax) specifically
> excludes quote marks from URIs, since they're a common URI delimeter in text
> documents and protocol fields.  What's more, both the "token" and
> "quoted-string" syntax elements come from a very different part of the SIP
> BNF; and no other URI syntax, as far as I know, uses anything like them.
> 
> I think it'd be much more in the spirit of the generic URI syntax to have 
> other-param be defined like this:
> 
>   other-param     = pname | pname "=" pvalue
>   pname           = 1*paramchar
>   pvalue          = 1*paramchar
>   paramchar       = param-reserved | unreserved | escaped
>   param-reserved  = "/" | "?" | ":" | "@" | "&" | "+" | "$" | ","
> 
> That is, special characters get %-escaped rather than quoted.
> "param-reserved" above is the same as the general "reserved" with ";" and
> "=", the parameter delimiters, omitted.
> 
> This would, however, be an incompatible change to the SIP spec.  What would
> people (especially the spec authors) think about this?
> 
> -- 
> Jonathan Lennox
> lennox@cs.columbia.edu
> 

Sounds reasonable. I doubt that there are many implementations that actually accept a
quoted-string as part of a SIP URL parameter value. And as you mention, in general
the URI syntax doesn't allow quotes and in practice I've never seem them used.

--
Sean Olson <sean.olson@ericsson.com>

> 
> _______________________________________________
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> 


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Subject: [SIP] Content Lenght in responses
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Why is the Content Length field not mandatory in cases of SIP responses ??. Wont
it simplify the message processing in case of TCP. The current rfc2543bis draft
says

"If a response does not contain a Content-Length, the client assumes that it
encompasses the remainder
of the UDP packet or the data until the TCP connection is closed, as applicable.
"

Please clarify ....

Natarajan




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Date: Tue, 02 May 2000 22:07:34 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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To: Sean Olson <eussean@exu.ericsson.se>
Cc: sip@lists.research.bell-labs.com, lennox@ober.cs.columbia.edu
Subject: Re: [SIP] Problem with the syntax of URI extension parameters
References: <200005030010.TAA17230@b04a45.exu.ericsson.se>
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Sean Olson wrote:
> 

> > I think it'd be much more in the spirit of the generic URI syntax to have
> > other-param be defined like this:
> >
> >   other-param     = pname | pname "=" pvalue
> >   pname           = 1*paramchar
> >   pvalue          = 1*paramchar
> >   paramchar       = param-reserved | unreserved | escaped
> >   param-reserved  = "/" | "?" | ":" | "@" | "&" | "+" | "$" | ","
> >
> > That is, special characters get %-escaped rather than quoted.
> > "param-reserved" above is the same as the general "reserved" with ";" and
> > "=", the parameter delimiters, omitted.
> >
> > This would, however, be an incompatible change to the SIP spec.  What would
> > people (especially the spec authors) think about this?
> >
> > --
> > Jonathan Lennox
> > lennox@cs.columbia.edu
> >
> 
> Sounds reasonable. I doubt that there are many implementations that actually accept a
> quoted-string as part of a SIP URL parameter value. And as you mention, in general
> the URI syntax doesn't allow quotes and in practice I've never seem them used.

Same here. Escaping the " kind of works, but is very unclean, since you
are now escaping syntactically relevant delimiters.

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


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Cc: dean.willis@wcom.com, sip@lists.research.bell-labs.com
Subject: Re: [SIP] Second try : Questions about REGISTER
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Sean Olson wrote:
> 
> > Personally, I wonder why we didn't just go ahead and include the user part
> > in the request-URI. This would allow my request-chaining REGISTER to be
> > fairly explicitly stated and solve lots of messiness.
> >
> > --
> > Dean
> >
> 
> I second that :) I've never understood the rationale behind excluding the user part in
> the Request-URI of a REGISTER request.
>

This may well be worth doing; the motivation was that REGISTER is
somewhat different in that you are not addressing the user directly, as
in INVITE. If a registrar/proxy server were to ignore that REGISTER is
special and needs to be handled locally, it would just send the request
to whatever foo@domain translates to.

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


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Subject: Re: [SIP] Problem with the syntax of URI extension parameters
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You would probably want to exclude "?" and "," from param-reserved: "?"
-  to eliminate the ambiguity with "?" that separates parameters from
headers, "," - because URLs are allowed in certain headers and allowing
commas in SIP URLs will make parsing those headers sometime impossible.

Another observation: to be as backward compatible as possible the
characters allowed in param-reserved should be a subset of those allowed
in token. The intersection of reserved from RFC2396 and token from
RFC2543 leaves us with the following

param-reserved = "+"

Also, note that unreserved in RFC2396 allows "(" and ")", while token
does not. On the other hand, I don't think that many implementations
will get broken if those extra characters are allowed.

---
Igor Slepchin


Jonathan Lennox wrote:
> I think it'd be much more in the spirit of the generic URI syntax to have
> other-param be defined like this:
> 
>   other-param     = pname | pname "=" pvalue
>   pname           = 1*paramchar
>   pvalue          = 1*paramchar
>   paramchar       = param-reserved | unreserved | escaped
>   param-reserved  = "/" | "?" | ":" | "@" | "&" | "+" | "$" | ","
>


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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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Cc: Jonathan Lennox <lennox@ober.cs.columbia.edu>,
        sip@lists.research.bell-labs.com
Subject: Re: [SIP] Problem with the syntax of URI extension parameters
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Igor Slepchin wrote:
> 
> You would probably want to exclude "?" and "," from param-reserved: "?"
> -  to eliminate the ambiguity with "?" that separates parameters from

Done.

> headers, "," - because URLs are allowed in certain headers and allowing
> commas in SIP URLs will make parsing those headers sometime impossible.

As far as I know, all headers where URLs are allowed mandate the use of
<> for URLs with commas or other "bad" characters. Any we missed?
(Probably still a good idea to make it reserved; the pain is small
compared to the confusion avoided.)

> 
> Another observation: to be as backward compatible as possible the
> characters allowed in param-reserved should be a subset of those allowed
> in token. The intersection of reserved from RFC2396 and token from
> RFC2543 leaves us with the following
> 
> param-reserved = "+"

Anybody object to this most restrictive coding? The only slight problem
with being restrictive is that it makes hand-coding HTML pages with SIP
URLs more painful. On the other hand, we haven't really defined any
parameters likely to see $ or @. However, removing : doesn't work well,
since we allow it in IPv6 numerical addresses in the maddr parameter.
Would seem strange to force escaping here.

> 
> Also, note that unreserved in RFC2396 allows "(" and ")", while token
> does not. On the other hand, I don't think that many implementations
> will get broken if those extra characters are allowed.
> 


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


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Date: Wed, 03 May 2000 00:33:00 -0400
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Subject: Re: [SIP] Content Lenght in responses
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Natarajan Kalpathy wrote:
> 
> Why is the Content Length field not mandatory in cases of SIP responses ??. Wont
> it simplify the message processing in case of TCP. The current rfc2543bis draft
> says
> 
> "If a response does not contain a Content-Length, the client assumes that it
> encompasses the remainder
> of the UDP packet or the data until the TCP connection is closed, as applicable.
> "
> 
> Please clarify ....

I believe this is the result of legacy HTTP, where Content-Length is not
mandatory in responses. I also believe this is because CGI scripts that
generate dynamic content wouldn't know the Content Length ahead of time,
so just sending data, and then closing the connection, was the easiest
way to determine message length. I can certainly envision placing of
dynamic content in SIP responses, so this may prove useful for us as
well.

Processing of responses without content length is pretty simple anyway.
It just means an additional counter that counts actual bytes read from
the response. When the connection closes, this value is your content
length.

-Jonathan R.



-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
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These are the hopefully final minutes of the SIP WG at the 47th IETF in
Adelaide.  They vary from the draft in that the consensus seems to be that
draft-campbell-sip-service-control-00.txt was not adopted as a WG item as
previously reported, but that the authors were encouraged to submit it as
an individual submission on informational track.

If there are no other changes I will conclude the revision of minutes and
post the final.

----

Minutes of SIP WG, IETF 47. Notes recorded by Tom Taylor.


1. Agenda Bashing
=================

The proposed agenda was accepted.  Presenters from Kyoto University
asked to be added to the agenda.  The request was denied because their
draft had not been submitted in time to be published by the IETF.

2. Status
=========

Dean Willis presented a summary of status of the WG work items,
followed by a brief discussion of potential charter items. There was
discussion as to whether the DCS drafts would be considered for
adoption by the WG. The current drafts still have RFC2026 complienace
issues preventing WG adoption as work items, but these issues may soon
be resolved. The general consensus seems that several of the DCS
drafts are likely to be adopted once the IPS is resolved.


The WG indicated consensus to adopt several tasks as chartered WG
activities, including:
SIP server features (standards track)
Call flows (informational)
Session Timer (standards track)
Reliability of provisional messages (standards track)
SIP-T efforts:
ISUP MIME types (standards)
ISUP mapping (undetermined)
Single Line Extension (undetermined)
Guidelines for Extensions.


As a general note regarding the management of the work of the WG:
Jonathan Rosenberg is attempting to assemble teams around specific work
items, with the proviso that individual participants should not be
over-committed.  The "single line extension" team is an example of this.



3. Summary of Current Discussion On The List
============================================

Jonathan Rosenberg summarized the status of discussion of a number of
current issues.

3.1 draft-nair-sip-dhcp-00.txt

No issues have been raised on the list, nor comments received from the
DHCP WG.  The fast-track procedure applied to this work appears to be
acceptable to the SIP WG.

3.2 Content In INVITE

Discussion in late December of how to include of content such as JPEG
images in INVITEs brought out a number of points:
 -- whether to include the actual content, include it indirectly through
provision of a URI, or include it indirectly as a media stream
 -- location of the content, in a header or in the body
 -- the need to identify how the content is to be used.
Jonathan perceived no consensus on these items, due to lack of comment
on specific proposals made to the list.


3.3 WWW-Authenticate Syntax

There is a proposal before the list, but no comments have been received.

3.4 tel:// vs. sip:// URLs

Jonathan emphasized the importance of the issue, discussion of which has
stalled.  He would like to see a new push to resolve it.

3.5 URL Parameters For Telephone Numbers

A couple of issues have been discussed under this heading: how to
indicate that the number shown is the result of an LNP lookup, and how
to indicate private dialing plans.  Again, Jonathan would like to see
a renewed effort to resolve these issues.

3.6 To Quote Or Not To Quote: Digest-URI

Robert Sparks commented that most implementations use quotes around the
digest-URI.  Usage for other fields is variable.  He suggests that all
fields be quoted.

3.7 Re-INVITE Glare

Jonathan noted that three proposals are on the table for handling of
coincident re-INVITEs.  Scott Petrack noted the need for a solution that
works with more than two parties.  Jonathan responded that SIP states
are shared only on a per-call-leg basis, except possibly in the case of
multicast, so multileg coincidental invites may not be an issue.

[A couple of points in Jonathan's charts were skipped because they are
being considered by the SIP security task force.]

3.8 When To Send 183 Response

Jonathan presented the rules for use of the 183 response which had come
out of discussion on the list.  A brief discussion followed, in which a
problem with the first rule was noted and resolved.

3.9 Overlap Dialling

Jonathan noted that there was no easy solution to the possibility of
forking for a call in which overlap dialling is being used.  David Oran
suggested that when the gateway rejects the first call, it should return
a contact which redirects all subsequent INVITEs to the same gateway.
Jonathan noted that this changed the meaning of the Contact: header
slightly, but its usage remains within acceptable bounds.

3.10 Early Media Criticality

Jonathan noted a continuing unresolved issue of how to distinguish
between critical and non-critical early media.

In general Jonathan intends to put out notes to the list to stimulate
further discussion of the open issues.

4. Status Of I-Ds (Jonathan Rosenberg)[charts]
======================================

4.1 Caller Preferences
 (draft-ietf-sip-callerprefs-01.txt)

Jonathan summarized a number of changes in the current draft.  There
have been no objections to the use of case-sensitive matching.  The
draft is due to go to the IESG, but Jonathan is looking for volunteeers
to check it out first.

4.2 Supported: Header
 (draft-ietf-sip-serverfeatures-02.txt)

The draft has undergone a major overhaul.  The one open point had to do
with use of the header with ACK.  This has been fixed and the draft has
been forwarded to the IESG.

4.3 Reliable Provisional Responses
 (draft-ietf-sip-100rel-00.txt)

A number of open issues still remain.
 -- cumulative acknowledgements: looking for an application to justify
inclusion of the mechanism
 -- response to PRACK: retain mechanism
 -- CANCEL for PRACK: disallow
We are looking to pass this document to the IESG as a Proposed Standard
by mid-April.

4.4 Guidelines For Authors Of SIP Extensions
 (draft-rosenberg-sip-guidelines-00.txt)


Jonathan proposed that this be a WG work item.  It would eventually
become a BCP, but not for another year.  For now it would be a living
document as the WG accumulates experience.  The sense of the meeting was
to adopt the work item, but there were mixed views on how quickly it
should be given formal status, with some arguing for near-term RFC
status.  This point will be taken to the list.  Jonathan has volunteered
to maintain the document if the WG chooses to make it a long-term
project as he proposes.

4.5 Third-Party Call Control
 (draft-rosenberg-sip-3pcc-00.txt)
This document has a number of dependencies, and various models of
control are possible.  In general the work does not involve an
extension to SIP, but only a description of the usage of its
mechanisms.  For this reason Jonathan does not propose that it be a WG
work item, but would like people to comment on it.

5. DCS Drafts
=============

5.1 Resource Reservation (Bill Marshall) [charts]
 (draft-manyfolks-sip-resource-00.txt)

This draft came about through the compromise merger of
draft-ietf-mmusic-qos-00.txt and draft-dcsgroup-sip-resource-01.txt.
The two-stage INVITE has been discarded.  SDP enhancements in support
of QOS and security negotiation include two new attributes (a=qos: and
a=security:), with syntax provided for confirmation that the requested
attribute values have been granted.  Extensions to SIP in support of
the negotiation of these attributes are covered in the same document
as the SDP changes, with AD approval. SIP extensions include use of a
header indicating that preconditions have been met.

The question was raised, whether there was any problem with using the
a=qos: attribute with provisional responses to allow the invitee to
set preconditions.  Bill made the point that the first user of the
attribute has to know whether the other end would understand it.

5.2 Caller Identity (Flemming Andreasen) [charts]
 (draft-dcsgroup-sip-privacy-01.txt)

Flemming summarized changes in the draft.  Earlier work has been
extended to recognize the possibility of boundaries between
proxies. In geneal, rather than requiring a nounary proxy to remove
calling party information, any proxy may encrypt the field and list
itself as an authenticator.  A new header, "Remote-Party-ID", has been
defined.  The set of possible operator services has been generalized.

Scott Petrack suggested that a specific tag for operator services might
not be in the spirit of SIP.  Jonathan agreed that an alternate approach
might be possible.  The basic idea, as always in SIP, is to define a
small set of reusable features rather than a variety of specialized
ones.

5.3 Distributing Call State (Bill Marshall) [charts]
 (draft-dcsgroup-sip-state-01.txt)

The mechanism has changed completely since the previous draft, in an
attempt to make it more general.  The draft now takes care to
distinguish between transaction state, connection state, and call
state.  The rules for including the "State" header have been
simplified.  There was a comment to the effect that the state
information must include the information necessary to ensure that it is
not used for a different call from the one which generated it.

5.4 Proxy-to-Proxy Extensions (Mike Mannette) [charts]
 (draft-dcsgroup-sip-proxy-proxy-01.txt)

Mike reviewed the changes from the previous draft.  LNP support has been
provided and hostport is now a required field in the "Gate" header.  The
"Billing-Info" and "Billing-Id" headers are unchanged.

5.5 Media Authorization (B. Beser) [charts]
 (draft-dcsgroup-sip-call-auth-01.txt)

This draft addresses the need for a per-call mechanism to authorize the
flow of media.  The new draft has been changed to focus only on the
client.  The former "DCS-Local-Gate" header has been renamed to
"Media-Auth".  The content has been generalized to hexadecimal.

6. Subscribe-Notify (Adam Roach)
==========================================
 (draft-roach-sip-subscribe-notify-00.txt)

Motivation: there is a need for a central definition of concepts which
have been mentioned in various documents.

The draft formalizes the methods and headers associated with
subscription and notification.  Distinctions are made between
call-related and resource-related subscriptions, with the latter
typically being of interest to third parties.  Rules are set down for
notifications and for duration of subscriptions.  The author attempted
to avoid disruption to PINT, but it might be possible to combine the
PINT work with this new material.

Scott Petrack noted that PINT had spent a great deal of time working
to make their mechanisms sufficiently generic.  He urged Adam to
review the PINT archives.  Unsubscription proved to be a tricky topic,
and PINT found it preferable to list events in bodies rather than in
headers.  Scott and Adam will discuss this further off-line.

Henry Sinnreich asked what measures had been taken to coordinate with
the Instant Messaging and Presence (IMPP) WG.  Jonathan replied that
IMPP is addressing a different problem and had already judged SIP to
be unsuitable for their purposes.  He also recalled that a general
subscription/notification BOF was held, but nothing came out of it
because the problem was too broad.

Scott Petrack noted that PINT will change if necessary to conform to
SIP.

An example subscription/notification service was presented, where the
caller is rung back when the callee becomes free.  There are open issues
where further work is required.  Jonathan proposed that another task
force be set up on the Egroup mail server, to refine the problem
statement.

Michael Thomas questioned whether SIP really needed an asynchronous
method of completing a call when the current protocol can support a
blocking approach.

7. Mobility (S. Baba) [charts]
=========================
 (draft-itsumo-sip-mobility-req-00.txt)

The draft provides requirements, open issues, and relevant work items
in SIP and other WGs.  The requirements are organized on a functional
basis.

Addressing the charts, Lawrence Conroy asked what was meant by the
requirement to "utilize" CDMA soft hand-off.  He saw this as invisible
at the IP level.  The presenter pointed out that the process might
involve multiple IP addresses for the same UA.  This would represent a
major extension to SIP.

James Polk noted that the forthcoming Location Information BOF might
be relevant to this problem.

[end of first session]

8. Review Of Charter Work Items  (Dean Willis) [charts]
===============================

Dean asked for comments on the proposed work items before the WG
chairs reviewed them and passed the results of the review on to the
list.  There were no comments.

9. INFO Method, Session Timer, 183 Session Progress (Steve Donovan)
[charts]
===================================================

9.1 INFO Method

Work on the INFO method, draft-ietf-sip-info-method-01.txt, is basically
done.

9.2 Session Timer

The Session Timer draft (draft-ietf-sip-session-timer-01.txt) underwent
significant change to reflect the new "Supported" header.  Some issues
remain before WG Last Call:
 -- refreshing of the timer.  Jonathan suggested that this be done by
re-INVITE, rather than a new method.  Agreed.
 -- Session-Expires header problematic in anything but INVITE -- will be
allowed with INVITE only.
 -- Can a proxy add Required headers? -- could end up with multiple
Required headers if the UAC also includes one.  This doesn't seem to
break anything, but there is an interaction with authorization.  In the
absence of comment from anyone but Jonathan, this should be checked out
on the list.
 -- the question was asked: what happens if the called UA includes
Session-Expires with a re-INVITE and it contradicts the one set by the
calling UA?  Steve suggested that it would be up to the implementation
to sort this out.  The list will be invited to comment.

Next steps:
 -- agreed that this becomes an official WG work item
 -- submit shortly to WG Last Call.

9.3 183 Session Progress

draft-ietf-sip-183-00.txt has a lot of dependencies, resulting in
improved understanding of the issues.  It may be used for early media
and for pre-conditions.  Summarizing the extensions involved:
 -- full session description bodies in 18x responses
 -- add the 183 response
 -- provide the "Session" header to indicate the purpose of the session
description body
The first two items are in 2543 bis already.

It is proposed that the Session header draft be refined to narrow its
focus, taking advantage of the 183 documentation.

10. Call Control (Robert Sparks) [charts]
================================

10.1 Framework For Call Control Extensions
 (draft-campbell-sip-cc-framework-00.txt)

Needs to be aligned with the guidelines.

10.2 Call Transfer
 (draft-sparks-sip-cc-transfer-00.txt)

Proposes a new TRANSFER method (not BYE ALSO), which does not alter the
original session.  Open issues:
 -- name of the method.  Jonathan interjected that he does not want to
build protocols for specific services, hence the method should have a
more abstract name and definition. Dean Willis suggested "Introduce".
 -- whether the method can carry bodies
 -- authentication.  Dean Willis suggested an "Introduced-By" header,
signed by the transferor.

10.3 Service Context -- Use of the SIP Request URI
 (draft-campbell-sip-service-control-00.txt)

This is an informational draft proposing the transmission of service
context through use of a distinctive Request-URI.  Robert provided an
example application and described the advantages of the mechanism.

David Oran was unclear on the intent of the work.  He noted that it
takes the opposite approach to that of HTML, by using the left side of
the URL rather than the right.  The syntax conflict could make the
result different depending on the server. Dean Willis pointed out that
SMTP also uses the left hand side, and that web URLs lack the @ and
therefore HAVE no left hand side.

Robert responded that the URL is intended to be an opaque entity bound
as a whole against a particular service.  David agreed that under these
conditions the format was acceptable.

The authors were encouraged to submit this work for informational
RFC status. This work was not adopted as a group effort of the
SIP WG.

10.4 Call Flow Examples (Presentd by Robert Sparks)
 (draft-ietf-sip-call-flows-00a.txt)

The draft has had a number of updates.  Please forward any further flows
to Alan Johnston (alan.johnston@wcom.com).

Jonathan noted that the document is getting too large to handle easily.
He proposes to farm out subsections to volunteers to review.  Ideally
they should simulate the flows to validate them.  Alan Johnston will
coordinate the work.

There is an issue with dependencies on work in progress.  The document
will be partitioned so that the stable part is processed first.

11. IETF SIP MIB (David Walker)
 (draft-ietf-sip-mib-00.txt)

Issue: the document is 70 pages long.  Can this be reduced?  It covers
RFC 2543 plus the INFO method and the 183 response.  Items are defined
within configuration, statistics, and notification groups for each of
the UAC, UAS, proxy, registrar, and redirect server, except that
elements common to all of these are grouped separately.  (Nothing has
been defined as yet to be specific to the redirect server.)

Concern: too many measurements.  Discussion of this brought out the
following:
 -- measurements should be defined only if there is a stated use for
them
 -- they are used as a troubleshooting tool, but a message history tool
would do a better job
David will send a note to the list to provoke further discussion.
Comments on how measurements are used will be welcome.  The note will
also raise a number of other questions for comment.

The current draft contains a couple of known errors.

12. BCP For SIP to ISUP Mapping (Gonzalo Camarillo) [charts]
 (draft-camarillo-sip-isup-bcp-00.txt)

This draft covers requirements, call flow examples, and the mapping of
parameters between the two protocols.  Gonzalo went over the changes to
the draft:
 -- new name
 -- additional extension: umbrella REQUIRE
 -- refinement of scope
 -- a discussion of SIP bridging between ISUP networks
 -- corrections to the handling of overlap signalling
 -- added examples.

He presented a walk-through of the overlap signalling case.  There is a
lot of messaging, but, unlike in the previous draft, it works.  There is
an open issue of how to control routing of overlap signalling in the
case of a stateless proxy forwarding to loadsharing MGCs.

A question was asked: how is connection redundancy handled -- for
example, UA multihoming.  What if the successive messages arrive at
different answers.  The reply was that this issue was out of scope of
the draft.  Jonathan remarked that SCTP transport for SIP is a topic
for the future.

Gonzalo concurrently presented a report on the status of the SIP-T
work.  It was noted that of the SIP-T work, the highest priority is to
obtain approval of the ISUP MIME types document. This is currently
planned for May 2000.

13. SIP Mapping Of MLPP (James Polk) [charts]
 (draft-polk-sip-mlpp-mapping-00.txt)

There is a mismatch between SIP "priority" levels and MLPP levels.
The draft proposals are to add a level and add administrative hooks.
Jonathan remarked that the standard is not the place to specify the
administrative hooks -- they belong in an RFP.

Subsequent discussion brought out two points:
 -- added priority level in RFC 2543bis
 -- additional document describing operation of MLPP.  This would cover
a number of items besides SIP, so it should not be a SIP WG document.

Jonathan also wondered whether MLPP really impacts SIP except at the
gateway, since it is concerned with preemption of limited call
resources in the face of higher-priority demand and IP resources are
not so obviously limited as circuits.  Brian Rosen disagreed, pointing
out that MLPP is not only about preeemption.

The outcome of the discussion was consensus to discuss adding another
priority level to 2543bis, or possibly adding "token" to the grammar
allowing arbitrary extension. This will require list review. James
Polk may submit a detailed fraft showing how the SIP priority levels
can be used to implement MLPP requirements.


14. Update On ISUP MIME Types (Lyndon Ong)
 (draft-ong-sip-qsig-mime-00.txt)

The draft has one open issue: the amount of ISUP version information
required. Followup is occuring in the SIP-T subteam and will be
reviewed on the main list when ready.

15. SIP and QOS (Henry Sinnreich) [charts]
 (draft-sinnreich-sip-qos-osp-01.txt)

Henry presented a list of "to do" items associated with the work.  He
hypothesized that a new WG may be required.  As evidence of this he
cited the need for a framework, within which different QOS models might
operate.

Radhika Roy mentioned that ITU-T Study Group 16 is also doing QOS work,
and the two groups could share their results.  Henry agreed that this
was possible.

Patrice Calhoun asked about the relationship between QOS and OSP (Open
Settlement Protocol, which appeared on Henry's charts).  Henry indicated
that OSP was relevant because it supports the clearing house model of
settlements, and the latter differs from the AAA WG model.

Melinda Shore remarked that overall (thinking also of DCS) there seemed
to be some strange QOS work going on, driven by charging models.  Hooks
within SIP are in scope for SIP WG discussion, but the general topic of
QOS for VoIP is out of scope.

Continuing in his presentation, Henry presented call flows showing the
tight coupling between signalling, QOS, and AAA.

Jonathan remarked again that ONLY changes required to SIP are within
the SIP WG's scope.

Dave Oran observed that DCS covers the "push model" of QOS: policy is
pushed to the enforcement point.  There is a need to develop a system
for the pull model, with open security issues standing in the
way. Dave proposed COPS-PR for this role.

Dean Willis suggested that perhaps the Transport Area WG could tackle the
larger problem.  Jonathan proposed discussing the matter with the ADs.

16. SIP Through Firewalls and NATs (Jonathan Rosenberg) [charts]
 (draft-rosenberg-sip-firewalls-00.txt)

Jonathan noted that the topic of firewall/NAT traversal is being covered
by the foglamps BOF.  The question is whether the SIP WG wants to do an
Informational RFC on the specific topic of SIP operation through these
barriers.

Dean Willis suggested that the question of whether this work belongs
in SIP is of the same order as whether MLPP does -- that is, both are
enablements for SIP within specific sectors.

There was a weak hum in favour of the work item.

17. Usage of TRIP For Gateway Registration (Jonathan Rosenberg)
 (draft-rs-trip-gw-00.txt)

This was simply an alert to the topic, which had been raised in the
meeting of the IPTEL WG the previous day.

18. Transferring User Control Information In SIP REGISTER Payloads
(Jonathan Rosenberg)
 (draft-lennox-sip-reg-payload-00.txt)

We need a way to pass scripts from the client to the server.  This
draft is the same as the original version a year ago, with minor
changes.  There are several open issues.  Jonathan asked "Should this
be a SIP work item?", which started a lively discussion.

Radhika Roy suggested there was a relationship to and potential impact
on mobility; Jonathan disagreed.  Lawrence Conroy wondered how the
scripts would be authenticated.  Dean Willis mentioned several
approaches that have been suggested.

Dean put the question: "even with the other alternatives, is it useful
to do this function via REGISTER?"  There was a slight hum in favour,
no opposition, with most registering no opinion.  The work was
accepted subject to approval on the list.

19. Finding A SIP Server With SLP (James Kempf)
 (draft-kempf-sip-findsrv-00.txt)

The intention is to add a template to the IANA service type registry.

Motivation: there are several ways to locate a SIP server through DNS --
for example, by use of a naming convention.  The method presented allows
location by characteristics.  This is useful because different proxies
in a domain can have different characteristics.  The approach taken is
to have one SLP abstract service type: "service-sip", with two concrete
service types.

Jim presented an example: find a registrar supporting CPL.

An open issue is how to handle incoming calls.  With other methods of
DNS usage, one can parse down the hierarchy of server names.  With SLP
it may be necessary to use SIP forking or multicast.

Should this be a WG work item or an individual draft?  IANA approval
requires RFC documentation and approval by Eric Guttman (servloc
Chair).  Comments on the draft can go to Jim directly or to the list.

Jonathan noted that this is one of many possible methods, but that's
OK.  There is certainly no point in discussing whether this work
should go forward as opposed to some other approach. The consensus was
to not adopt this as a WG task at this time, but to follow the work as
it matures and reconsider at a futue point.

20. CGI Interface (Jonathan Rosenberg)
 (draft-lennox-sip-cgi-03.txt)

The draft is being cleaned up.  A couple of issues have been resolved.
There are alternative methods for transfer of the CGI scripts. There
appears to be no intent to make this a WG item at this time.

21. Distributed Multipoint Conferences Using SIP (Jeff Mark) [charts]
 (draft-mark-sip-dmcs-00.txt)

Jeff provided some examples, showing manipulation of full mesh
conferences.  Is there SIP interest in further such full mesh work?

Jonathan noted that conference control is not part of the SIP WG mandate.
The problem of topology management is overly complicated because call
control should be separated from link state propagation.  The latter is not
something SIP is good at.

Jonathan will organize a first pass at rearchitecting the proposal,
but there are other approaches to multi-party conferencing that may prove
worth discussing.


--
Dean Willis
dwillis@greycouncil.com








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Date: Wed, 03 May 2000 00:45:12 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Cc: Sean Olson <eussean@exu.ericsson.se>, dean.willis@wcom.com,
        sip@lists.research.bell-labs.com
Subject: Re: [SIP] Second try : Questions about REGISTER
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Henning Schulzrinne wrote:
> 
> Sean Olson wrote:
> >
> > > Personally, I wonder why we didn't just go ahead and include the user part
> > > in the request-URI. This would allow my request-chaining REGISTER to be
> > > fairly explicitly stated and solve lots of messiness.
> > >
> > > --
> > > Dean
> > >
> >
> > I second that :) I've never understood the rationale behind excluding the user part in
> > the Request-URI of a REGISTER request.
> >
> 
> This may well be worth doing; the motivation was that REGISTER is
> somewhat different in that you are not addressing the user directly, as
> in INVITE. If a registrar/proxy server were to ignore that REGISTER is
> special and needs to be handled locally, it would just send the request
> to whatever foo@domain translates to.

I'm not sure I see what is gained by adding a user portion. What is the
semantic of having a user portion in the request URI? This still doesn't
change how a proxy looks up registrations when an INVITE request is
received (still based on request URI), which is what Dean is asking for.

I agree completely with Dean that there is value in being able to route
based on the To field for certain things. It is my belief that out of
the ordinary routing and request handling services, in general, will
require programmability in proxies (though CPL, or more likely CGI or
servlets). I can think of a reason to making a routing decision based on
almost any header field in the request. Rather than specifying these
things in the protocol, leaving them to custom applications gives you
maximum flexibility.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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        sip@lists.research.bell-labs.com
Subject: Re: [SIP] Problem with the syntax of URI extension parameters
References: <200005022203.SAA34631@conrail.cs.columbia.edu> <390F9700.F3FA43D8@dynamicsoft.com> <390F9E8E.10FAEAB1@cs.columbia.edu>
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Henning Schulzrinne wrote:
> 
> Igor Slepchin wrote:
> >
> > You would probably want to exclude "?" and "," from param-reserved: "?"
> > -  to eliminate the ambiguity with "?" that separates parameters from
> 
> Done.
> 
> > headers, "," - because URLs are allowed in certain headers and allowing
> > commas in SIP URLs will make parsing those headers sometime impossible.
> 
> As far as I know, all headers where URLs are allowed mandate the use of
> <> for URLs with commas or other "bad" characters. Any we missed?
> (Probably still a good idea to make it reserved; the pain is small
> compared to the confusion avoided.)

I agree the , should be escaped to avoid confusion.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Wed May  3 06:22:11 2000
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From: Ge Yu <engp9374@nus.edu.sg>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Shail Bhatnagar <shbhatna@cisco.com>,
        "'Alexandru Gavrilescu'" <alexang@Exchange.Microsoft.com>
Cc: SIP IETF <sip@lists.research.bell-labs.com>
Subject: RE: [SIP] Second try : Questions about REGISTER
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Hi, friends,

As I know through 16.1 of RFC 2543, the registration process is as follows:

1. Watson firstly sends the REGISTER request to the server at bell-tel.com,
with Contact: Watson@example.com. That is, he hasn't contacted with the
server at example.com ahead of REGISTER. 
    So what happens if another Watson@example.com has been registered in
example.com domain?

2. Watson registers to the server at example.com, which confirms the
registration, through the process the server at  example.com knows where to
forward the INVITE to Watson.

3. the server at example.com contacts with Registar( or it acts as Registar
itself), and stores the updated URL in Location server.

4. the sender's proxy server enquiries the location server, knows the
current URL of recepient and forwards the INVITE to example.com, which
transfers it to Watson. 

Anything missed?

Very appreciated with your kindly feedback.

Ge Yu

> ----------
> From: 	Alexandru Gavrilescu[SMTP:alexang@Exchange.Microsoft.com]
> Sent: 	Wednesday, May 03, 2000 2:48 AM
> To: 	Jonathan Rosenberg; Shail Bhatnagar
> Cc: 	SIP IETF
> Subject: 	RE: [SIP] Second try : Questions about REGISTER
> 
> 
> I have a question regarding Jonathan's answer to the Shail's third
> question. 
> Do we assume that tawatson has an account created for himself at
> example.com or he is just a visitor at example.com? 
> I assume the first. 
> What would be the behaviour is tawatson@bell-tell.com would be just a
> visitor at example.com? 
> 
>   
> -----Original Message----- 
> From: Jonathan Rosenberg [ mailto:jdrosen@dynamicsoft.com] 
> Sent: Monday, May 01, 2000 9:58 PM 
> To: Shail Bhatnagar 
> Cc: SIP IETF 
> Subject: Re: [SIP] Second try : Questions about REGISTER 
> 
> 
> 
> 
> Shail Bhatnagar wrote: 
> > 
> > In section 16.1 of RFC2543bis there is an example of Registration. I 
> > have 
> > a couple of questions about this scenario. I would appreciate any 
> > responses. 
> > 
> > 1) What would be the To header of the REGISTER that T. Watson sends to 
> > the registrar at example.com. Would it be tawatson@example.com or 
> > tawatson@bell-tel.com ?? As per the description in section 4.2.6, To 
> > header 
> > domain could be different from the domain of the Request-URI in the 
> > REGISTER. 
> 
> Most likely tawatson@example.com. This is because if some UAC or proxy 
> wants to send a request to tawatson@example.com, it will use DNS to find 
> the SIP server for example.com, and forward the request there. As a 
> result, watson will want to register tawatson@example.com with the 
> server for which requests to example.com will be routed. The spec does 
> not prohibit registering tawatson@bell-tel.com with example.com. 
> However, example.com is not likely to receive requests with domains of 
> bell-tel.com in the request URI. 
> 
> > 
> > 2) Let us say x@bell-tel.com (From: x@bell-tel.com) sends an INVITE to 
> > the 
> > proxy at bell-tel.com and the Request-URI is INVITE watson@bell-tel.com.
> 
> > As per the latest registration, the Contact is tawatson@example.com, so 
> > the proxy at bell-tel.com rewrites the Request-URI and sends it to the 
> > proxy at example.com - INVITE tawatson@example.com. Now what is the 
> > search 
> > key at the registrar - is it user@domain or simply user. If it is simply
> 
> > user, 
> > then domain of To header in REGISTER is meaningless. If it is 
> > user@domain, 
> > then To header of the REGISTER sent to the registrar at example.com must
> 
> > be 
> > tawatson@example.com. 
> 
> Thats your choice. If your proxy only handles a single domain, using 
> user alone as a lookup key into your database is fine. If your proxy 
> handles multiple domains, you will want to use the complete user@host. 
> 
> 
> > 
> > 3) What if there already exists a user tawatson at the domain 
> > example.com. 
> > Should the registrar at example.com reject the registration ?? If yes, 
> > what response code should be used ?? 
> 
> No, the new Contact would be added to the existing registration. 
> 
> -Jonathan R. 
> 
> -- 
> Jonathan D. Rosenberg                       72 Eagle Rock Ave. 
> Chief Scientist                             First Floor 
> dynamicsoft                                 East Hanover, NJ 07936 
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778 
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244 
> http://www.dynamicsoft.com 
> 
> 
> _______________________________________________ 
> SIP mailing list 
> SIP@lists.bell-labs.com 
> http://lists.bell-labs.com/mailman/listinfo/sip 
> 
> 


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From: Pedro Pedrosa <Pedro.Pedrosa@lis2.siemens.pt>
To: Peter Blatherwick <blather@nortelnetworks.com>,
        "'hiren.shah@wipro.com'" <hiren.shah@wipro.com>,
        sip <sip@lists.bell-labs.com>
Subject: RE: [SIP] SIP/MGCP
Date: Wed, 3 May 2000 11:55:29 +0100 
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Hello,
 
Is there any mailing list like the one we have for SIP? It would be highly
interesting!
 
How can I join?
 
 
Best Regards
Pedro Pedrosa

-----Original Message-----
From: Peter Blatherwick [mailto:blather@nortelnetworks.com]
Sent: Terça-feira, 25 de Abril de 2000 15:11
To: 'hiren.shah@wipro.com'; sip
Subject: RE: [SIP] SIP/MGCP



Hiren: 

Please join the Megaco WG discussion, since that is the IETF Standards Track
direction for gateway control.  Megaco/H.248 Protocol is the international
standard in this area, a joint effort between IETF Megaco and ITU SG-16.  

WG Charter, list instructions etc can be found at: 
http://www.ietf.org/html.charters/megaco-charter.html
<http://www.ietf.org/html.charters/megaco-charter.html>  

E-mail list archive is at: 
http://standards.nortelnetworks.com/archives/megaco.html
<http://standards.nortelnetworks.com/archives/megaco.html>  

Regards, 
Peter Blatherwick 


-----Original Message----- 
From: hiren.shah@wipro.com [ mailto:hiren.shah@wipro.com
<mailto:hiren.shah@wipro.com> ] 
Sent: Tuesday, April 25, 2000 08:57 
To: sip 
Subject: [SIP] SIP/MGCP 


Hi all 
nara and hemant have really cleared my doubts regarding the use of SIP in 
MGCP 
protocol.Thanks a lot. 
This is a bit off the line but can anyone direct me to a MGCP discussion 
forum. 
Basically i am doing some research in MGCP and SIP. 

regards 
Hiren 






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From: Pedro Pedrosa <Pedro.Pedrosa@lis2.siemens.pt>
To: Pedro Pedrosa <Pedro.Pedrosa@lis2.siemens.pt>,
        Peter Blatherwick <blather@nortelnetworks.com>,
        "'hiren.shah@wipro.com'" <hiren.shah@wipro.com>,
        sip <sip@lists.bell-labs.com>
Subject: RE: [SIP] SIP/MGCP
Date: Wed, 3 May 2000 11:58:41 +0100 
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Hello,

Forget it! I have found out how!


Best Regards
Pedro Pedrosa

		-----Original Message-----
		From:	Pedro Pedrosa [mailto:Pedro.Pedrosa@lis2.siemens.pt]
		Sent:	Quarta-feira, 3 de Maio de 2000 12:55
		To:	Peter Blatherwick; 'hiren.shah@wipro.com'; sip
		Subject:	RE: [SIP] SIP/MGCP

		Hello,
		 
		Is there any mailing list like the one we have for SIP? It
would be highly
		interesting!
		 
		How can I join?
		 
		 
		Best Regards
		Pedro Pedrosa

		-----Original Message-----
		From: Peter Blatherwick [mailto:blather@nortelnetworks.com]
		Sent: Terça-feira, 25 de Abril de 2000 15:11
		To: 'hiren.shah@wipro.com'; sip
		Subject: RE: [SIP] SIP/MGCP



		Hiren: 

		Please join the Megaco WG discussion, since that is the IETF
Standards Track
		direction for gateway control.  Megaco/H.248 Protocol is the
international
		standard in this area, a joint effort between IETF Megaco
and ITU SG-16.  

		WG Charter, list instructions etc can be found at: 
		http://www.ietf.org/html.charters/megaco-charter.html
		<http://www.ietf.org/html.charters/megaco-charter.html>  

		E-mail list archive is at: 
		http://standards.nortelnetworks.com/archives/megaco.html
		<http://standards.nortelnetworks.com/archives/megaco.html>  

		Regards, 
		Peter Blatherwick 


		-----Original Message----- 
		From: hiren.shah@wipro.com [ mailto:hiren.shah@wipro.com
		<mailto:hiren.shah@wipro.com> ] 
		Sent: Tuesday, April 25, 2000 08:57 
		To: sip 
		Subject: [SIP] SIP/MGCP 


		Hi all 
		nara and hemant have really cleared my doubts regarding the
use of SIP in 
		MGCP 
		protocol.Thanks a lot. 
		This is a bit off the line but can anyone direct me to a
MGCP discussion 
		forum. 
		Basically i am doing some research in MGCP and SIP. 

		regards 
		Hiren 






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From sip-admin@lists.bell-labs.com  Wed May  3 11:12:21 2000
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Date: Wed, 03 May 2000 11:10:34 -0400
From: "Mathieu Gervais (LMC)" <lmcmatg@lmc.ericsson.se>
Organization: LMC, Ericsson Research Canada
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Subject: [SIP] SIP URL support in Mozilla : sip not official URL scheme?
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Hi all,

 I just posted a Request For Enhancement in Bugzilla, the Mozilla bug
database.
See http://bugzilla.mozilla.org/show_bug.cgi?id=37637

Comments about the RFE where added, from a Mozilla developer, saying SIP
"is not on the list" of official URL schemes
 @  ftp://ftp.isi.edu/in-notes/iana/assignments/url-schemes (see
http://www.iana.org/numbers.htm#U).

Why isn't sip there? Is the registration of the sip url scheme with IANA
"in process"? 

See also "Registration Procedures for URL Scheme Names" @
http://www.rfc-editor.org/rfc/bcp/bcp35.txt

Feel free to add comments in bugzilla, but keep in mind it's a bug
database, not a discussion forum.

Thanks, 


 Mathieu


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From sip-admin@lists.bell-labs.com  Wed May  3 11:18:13 2000
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Date: Wed, 3 May 2000 08:19:33 -0700 (PDT)
From: James Kempf <James.Kempf@Eng.Sun.COM>
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Subject: [SIP] Applicability of IP Mobility to CDMA Soft Handoff
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--Sounder_of_Swine_044_000
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There has been some discussion on the mobile IP and SIP lists about
replacing CDMA soft handoff with IP mobility mechanisms. The attached draft, 
authored by Pete McCann, Phil Roberts, and myself, attempts to explain how
a CDMA RAN works and why soft handoff is not a promising candidate
for replacing with IP mobility mechanisms in the near term. In contrast,
the analysis is not applicable to hard handoff, which is a potential
candidate for replacing with IP mobility. We have submitted this to
the drafts alias as an individual contribution 

Comments are welcome.

		jak

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INTERNET DRAFT                                                James Kempf
Category: Informational                             Sun Microsystems, Inc.
Title: draft-kempf-cdma-appl-00.txt                           Peter McCann
Date: April 2000                                       Lucent Technologies
                                                            Philip Roberts
                                                             Motorola, Inc.


            IP Mobility and the CDMA Radio Access Network:
                Applicability Statement for Soft Handoff



Status of this Memo

   This document is an individual contribution for consideration by the
   Mobile IP Working Group of the Internet Engineering Task Force.

   Distribution of this memo is unlimited.

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.  Internet-Drafts are working
   documents of the Internet Engineering Task Force (IETF), its areas,
   and its working groups.  Note that other groups may also distribute
   working documents as Internet-Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   The list of current Internet-Drafts can be accessed at:

      http://www.ietf.org/ietf/1id-abstracts.txt

   The list of Internet-Draft Shadow Directories can be accessed at:

      http://www.ietf.org/shadow.html.

   Copyright   (C) The Internet Society 2000.  All Rights Reserved.











Kempf, McCann, Roberts    expires October 2000                  [Page 1]

INTERNET DRAFT                                                April 2000


Abstract

   Recently, there have been a variety of proposals submitted to the
   Mobile IP Working Group and to other IETF working groups for IP
   mobility solutions that seek to enhance or replace mobile IP. These
   proposals, often characterized as micromobility or fast handoff, are
   addressed primarily at the perceived need of multimedia sessions such
   as video or voice over IP for faster handoff between radio base
   stations, and are primarily directed at real time multimedia traffic
   in 3rd generation cellular access networks. In this paper, we discuss
   the design of CDMA radio access networks (RANs) and the applicability
   of IP mobility to soft handoff in a CDMA RAN. We attempt to show that
   given current IP routing algorithms and the constraints on a CDMA
   RAN, IP mobility solutions have little, if any, role to play in
   handoff within the RAN. In contrast, an IP mobility solution is
   likely to play a big role in fast handoff between RANs, also called
   hard handoff. While future developments in IP networking may change
   this situation, IP mobility in CDMA networks currently seems to apply
   only when the mobile node roams between RANs rather than between base
   stations within a RAN.

Table of Contents

   1.0  Introduction
   2.0  Terminology
   3.0  RAN Architecture and Characteristics
   4.0  Applicability of IP Mobility to Soft Handoff
   5.0  Applicability of IP Mobility to Hard Handoff
   6.0  Future Prospects for IP Mobility in the CDMA RAN
   7.0  Summary
   8.0  References
   9.0  Authors' Addresses
   10.0 Full Copyright Statement


1.0  Introduction

   Mobile IP [1] allows IP hosts that change their point of attachment
   to the network to keep their IP address as they change from their
   home subnet to other subnets. Recently, there have been a variety of
   proposals advanced for augmenting or replacing mobile IP in access
   networks for cellular telephony systems. These proposals are often
   characterized as supporting micromobility or fast handoff, and are
   directed towards real time multimedia streams in 3rd generation
   cellular networks (see [2] [3] [4] [5] and [7]).

   While these proposals may have some applicability if handoff between
   RANs is very frequent, their utility is lessened in the presence of



Kempf, McCann, Roberts    expires October 2000                  [Page 2]

INTERNET DRAFT                                                April 2000


   link-layer mobility like that offered by today's TDMA and CDMA
   systems.  In fact if the link-layer offers transparent mobility
   throughout a given domain these proposals do not contribute anything
   since they are essentially intra-domain mobility management
   protocols.  As cellular networks evolve towards more pervasive IP
   technologies, mobility for traffic within those networks must
   accommodate some level of movement of IP traffic.

   In this paper, we discuss CDMA radio access networks and why IP
   mobility solutions (including mobile IP) are not applicable to soft
   handoff in a CDMA radio access network. In effect, the CDMA RAN
   network with soft handoff is an application layer transport and
   mobility mechanism, and therefore not an appropriate candidate for
   moving into the network layer. In contrast, IP mobility solutions
   that are directed at improving handoff performance within the core
   network, often called hard handoff, are likely to play an important
   role in enhancing the performance of IP networking for CDMA (see [6]
   for an example).

2.0  Terminology

   mobile terminal
     A mobile IP host. In mobile IP terminology, this is called the
     mobile node.

   base station
     A fixed, land-based radio transmitter and receiver, used to provide
     cellular telephony radio coverage in a limited geographic area.  A
     mobile terminal may be in contact with one or more base stations at
     a time in CDMA networks. Also called the Base Transceiver Station
     (BTS) or Node B.

   RAN
     The radio access network. This is a wired network that sits between
     a collection of base stations and the core, wired telephone
     network.  The RAN in CDMA systems is involved in real-time
     distribution and collection of physical layer radio frames to and
     from base stations, a topic that is discussed in the next section.

   soft handoff
     The process by which a moving CDMA mobile terminal is transferred
     between one base station or set of base stations to another within
     the radio access network. Soft handoff is typically very fast (on
     the order of 20 ms) and has a low probability of dropping ongoing
     real time connections.

   hard handoff
     The process by which a moving mobile terminal is transferred



Kempf, McCann, Roberts    expires October 2000                  [Page 3]

INTERNET DRAFT                                                April 2000


     between one cellular service provider's network and another or
     between two RANs that do not share a direct connection within a
     provider's network.  Hard handoffs have a higher probability of
     connection droppage, and are slower (usually 100 ms or more).

   macrodiversity
     A term used to describe the fact that within the CDMA RAN, there is
     no single octet stream corresponding to the data that arrives at or
     is sent by the mobile terminal. Macrodiversity results because the
     mobile terminal can be in contact with more than one base station
     at a time.

   frame selector
     A combination software/hardware unit at the gateway to the RAN that
     combines the multiple octet streams from multiple base stations in
     contact with a single mobile terminal into a single octet stream. A
     similar process happens at the mobile terminal. Also called the
     macrodiversity combiner or Selection and Distribution Unit (SDU).

   RAN gateway
     A functional unit positioned between the RAN and the core network.
     The RAN gateway includes the frame selector, in addition to
     functional units that perform soft handoff and radio frame
     processing.  Also called the Base Station Controller (BSC) or Radio
     Network Controller (RNC).

   radio frame
     A short (usually 20 millisecond) unit of transmission at the
     physical radio layer used to transmit data over the air to and from
     the mobile terminal.  The frames do not contain complete IP
     packets, but rather contain small sections of octet stream data
     that must be framed by a higher layer protocol (such as PPP) to
     form IP packets.  On a basic fundamental data rate channel one
     radio frame contains about 20 octets.  Radio frames may be
     retransmitted a small number of times to increase the reliability
     of the octet stream transport.  This is performed by a negative-
     acknowledgement protocol known as the Radio Link Protocol (RLP).

3.0  RAN Architecture and Characteristics


   A RAN consists of a RAN gateway connected to one or more base
   stations.  In the network to mobile terminal (forward) direction, the
   RAN gateway performs the following functions:


     1) Receive packets from the core network destined to the mobile
     terminal,



Kempf, McCann, Roberts    expires October 2000                  [Page 4]

INTERNET DRAFT                                                April 2000


     2) Process those packets into radio frames,

     3) Replicate the radio frames and transmit copies to base stations
     that are currently in contact with the mobile terminal in a way
     such that the frames arrive at each base station in a timely
     fashion,

     4) Manage the retransmission of individual radio frames when
     negative acknowledgements are received.


   In the mobile terminal to network (reverse) direction, the RAN
   gateway performs the following functions:


     1) Collect copies of the radio frames fowarded by base stations
     that are currently in contact with the mobile terminal,

     2) Combine these (possibly errorful) copies into one (hopefully
     error-free) radio frame,

     3) From the resulting radio frame stream, synthesize an outgoing
     octet stream of packets for the core network.


   In both directions, the RAN gateway performs the following function:


     1) Manage the power with which mobile nodes and base stations are
     transmitting, so as to maintain a low error rate while at the same
     time minimizing the transmitted power and therefore interference
     among different transmitters (and drain on the mobile terminal's
     battery).

     2) In concert with the mobile terminal, manage the set of base
     stations with which a mobile terminal is in contact such that the
     quality of the radio signal is maintained as the mobile terminal
     moves.


   The use of multiple base stations to transmit and receive the same
   radio frames to and from the mobile node at the same time is known as
   "macrodiversity" and can help to improve the reliability of the
   wireless link.  Note that some of the base stations may be owned by a
   neighboring RAN and this necessitates a RAN-to-RAN interface to carry
   the radio frames.

   The following figure illustrates the architecture and how the CDMA



Kempf, McCann, Roberts    expires October 2000                  [Page 5]

INTERNET DRAFT                                                April 2000


   RAN network works:

                         Core Network

                            ^
                            |   incoming/outgoing octet stream
                            |    from/to corresponding node (129.142.68.79)
                            |
                            V
   +-----------------------------------------------------+
   | RAN Gateway                                         |
   |                                                     |
   |    +-------------+  +-----------+  +----------+     |
   |    |             |  |           |  |          |     |
   |    | Frame       |  | Frame     |  | Soft     |     |-----------
   |    | Selection   |  | Splitting |  | Handoff  |     |       to another
   |    |             |  |           |  | Control  |     |       RAN Gateway
   |    +-------------+  +-----------+  +----------+     |
   |                                                     |
   |    +---------------+                                |
   |    |               |                                |
   |    | Radio Frame   |                                |
   |    | Processing    |                                |
   |    |               |                                |
   |    +---------------+                                |
   |                                                     |
   +-----------------------------------------------------+
            |             |              |            |    *
            |             |              |            |      *
            |             |     ...      |            |       * radio
            |             |              |            |       * frames
            |             |              |            |      *
            |             |              |            |    *
       +---------+   +---------+   +---------+   +---------+
       | Base    |   | Base    |   | Base    |   | Base    |
       | Station |   | Station |   | Station |   | Station |
       |   B1    |   |   B2    |   |  B42    |   |   B43   |
       +---------+   +---------+   +---------+   +---------+
            \              |
             \             V
              \--------->
                           |
                         -----
                        /     \
                    -----       -----           162.42.42.42
                   | mobile terminal|(  --->
                   ------------------
                     ( )        ( )



Kempf, McCann, Roberts    expires October 2000                  [Page 6]

INTERNET DRAFT                                                April 2000


   The links between the RAN gateway and the base stations are typically
   point to point links today, though provisions exist in the 3rd
   generation standards for switched networks.

   In the above figure, a mobile terminal with IP address 162.42.42.42
   is corresponding with a host having the IP address 129.142.68.79,
   through a CDMA cellular network. The mobile terminal is in contact
   with two base stations, B1 and B2. The RAN gateway takes incoming
   packets from 129.142.68.79 and splits them into two streams that it
   sends to base stations B1 and B2. A mobile terminal can be in
   communication with up to 3 and, in some CDMA systems, up to 6 base
   stations at a time. When the multiple octet streams are received at
   the mobile terminal, the mobile terminal performs a sophisticated
   combination of the multiple packet streams at L1 to deliver the end
   packet to the application.

   Packets flowing in the other direction, from 162.42.42.42 to
   129.142.68.79, are put into an octet stream which is then divided
   into radio frames.  Each radio frame is received by B1 and B2 and
   delivered to the frame selector in the RAN gateway. The frame
   selector performs signal processing on the incoming radio frames and
   produces a single frame that is then sent to the re-sequencing buffer
   where the octet stream is re-created.  The IP packets are formed from
   the octet stream and transmitted into the core IP network.


   An important point to note about the RAN is that, even if the RAN
   itself is running IP, routing in the RAN does not use the mobile's IP
   address. In fact, the IP packets sent by the mobile terminal are not
   tunneled through the RAN nor do they necessarily appear in a form
   that would be recognizable using a packet sniffer. The packets in the
   RAN contain radio frames that have been highly processed into a form
   that is extremely efficient for the base stations to handle and that
   efficiently uses radio spectrum, including compensations for the
   inherently lossy nature of the radio medium.

   On the forward leg to the mobile terminal, because the delay and
   jitter constraints between multiple base stations transmitting to the
   same mobile terminal are so tight (5ms to 80ms), the RAN gateway
   essentially puts out streams of radio frames that the base station
   can quickly pull off the wire and transmit over the air. On the
   reverse leg from the mobile terminal, the base station simply pulls
   the radio frames off the air and puts them on the wire without any
   further processing. The RAN gateway's radio frame processor is
   responsible for making sure that the jitter and delay constraints are
   met, and for processing packets from the core network into radio
   frames.




Kempf, McCann, Roberts    expires October 2000                  [Page 7]

INTERNET DRAFT                                                April 2000


   When the mobile terminal begins to move out of range of stations B1
   and B2, the RAN gateway adds and deletes base stations from the set
   currently serving the mobile node.  This process is called soft
   handoff.  Note that new base stations may belong to a neighboring RAN
   which requires closely-coupled RAN-to-RAN interaction. The real time
   constraints on soft handoff are extremely tight. All base stations
   involved in soft handoff must transmit the command for the mobile to
   move at the same time. North American cellular networks use the
   Global Positioning System as a time source to assure that these
   timing constraints are met.

   As shown in the figure, two RAN gateways can be connected together
   through a direct link. This link allows radio frames containing data
   and RAN control protocol to flow between two RANs. As a result, two
   RANs can perform soft handoff between them, increasing the quality
   and reliability of the connection when a user moves between coverage
   areas. A RAN gateway and its collection of base stations can only
   cover a limited geographic area, so RAN interconnection is very
   important in cellular networks for maintaining good connection
   quality over large geographic areas.

4.0 Applicability of IP Mobility to Soft Handoff

   Most of the proposals for IP mobility in the RAN assume the
   following:


     1) An end-to-end IP routing model for routing through the RAN.

     2) A one-to-one mapping between the mobile terminal and the base
     station with which it is in contact.

     3) The IP packets coming from the mobile terminal are transported
     directly on L2 in the RAN.


   As the above discussion has attempted to show, the extremely tight
   real time constraints on traffic in the RAN require that RAN traffic
   be highly processed as radio frames for efficient delivery into and
   from the radio medium by the base stations. This precludes routing IP
   packets from the mobile directly over the RAN. Furthermore, because
   there are multiple octet streams flowing over the RAN that correspond
   to one logical octet stream going to and from the mobile terminal,
   there is no one-to-one mapping between the mobile terminal and a base
   station.

   Taking mobile IP as an example, using mobile IP in the RAN would
   require that a mobile IP foreign agent (FA) be present at each base



Kempf, McCann, Roberts    expires October 2000                  [Page 8]

INTERNET DRAFT                                                April 2000


   station. However, because the mobile terminal can be in contact with
   up to 6 base stations at once, there is no unique care-of address for
   the mobile (unless the mobile uses a co-located care-of address).
   Therefore, the home agent (HA) would need to forward packets to
   multiple FAs.  Furthermore, on the reverse leg, a frame selector is
   still necessary to generate a single packet for the corresponding
   node.

   In effect, the RAN controller is an application level transport and
   mobility mechanism specialized to the CDMA radio medium. It is
   therefore not a good candidate for replacing with network level
   mobility mechanisms.  Because of the hard real time constraints
   involved in soft handoff, the RAN controller's soft handoff function
   is also not a good candidate for replacing with more general
   application level mechanisms, such as SIP [4].

5.0 Applicability of IP Mobility to Hard Handoff

   Note that the above considerations do not apply to hard handoff,
   which occurs outside of the RAN. When a mobile terminal moves between
   two RANs that are not interconnected, the RAN controller defers
   handoff to the core network. Some mechanism is necessary in the core
   network to move the mobile terminal's point of attachement at the
   network level. IP mobility solutions for fast handoff are applicable
   here.

   Proposals such as [6] that apply after frame selection do not involve
   soft handoff and therefore are appropriate for implementing fast,
   hard handoff.

6.0 Future Prospects for IP Mobility in the CDMA RAN

   What would it take to enable IP mobility in the RAN? The question is
   worth examining because the end-to-end model of networking which
   would be required to make IP mobility work in the RAN has attractions
   from the point of view of simplicity of network management and
   transparency.

   Certainly, routing the mobile's packets directly in the RAN would be
   a major contributor. For that to happen, IP over the air interface
   would be necessary. The major impediment to IP over the air is
   currently header size, but new work in header compression may
   eliminate this objection. Given that a spectrally efficient
   representation of IP on the radio medium is possible, IP packets can
   be sent out over the air by the mobile terminal, and the base
   stations can handle the packets precisely as they currently do with
   the specialized radio frames.  The result would be that IP packets
   from the radio would appear directly on L2 in the RAN.



Kempf, McCann, Roberts    expires October 2000                  [Page 9]

INTERNET DRAFT                                                April 2000


   However, there is still a problem with macrodiversity. On the forward
   leg, the routing from a single source at the RAN gateway to multiple
   base stations looks like multicast, but the real time constraints are
   extremely tight. A multicast routing algorithm with routing tree
   modifications that converge in real time might contribute to solving
   the problem. On the other hand, the reverse leg traffic would still
   need to be combined in the wired network behind the base stations.

   While the prospects for moving IP into the RAN are good, it seems
   unlikely that IP mobility will play any role in replacing CDMA soft
   handoff in the immediate future. CDMA soft handoff appears to be a
   very specialized form of transport-level or link layer mobility, and
   therefore not a good candidate for replacement by more general
   network or application level mechanisms. In addition, even if IP is
   used for transport in the RAN, if voice over IP is to completely
   replace the current radio voice protocols, voice packets may need to
   be processed at the RAN gateway for maximum efficiency and robustness
   over the radio medium.

7.0 Summary

   Most proposals for IP mobility require a one-to-one mapping between
   the mobile terminal and a base station with which it is in contact,
   and assume end-to-end connectivity and consequently routing in the
   radio access network based on the mobile's IP address. In CDMA
   networks, macrodiversity and the need for extremely low delay and
   jitter invalidate these assumptions. Consequently, IP mobility
   solutions are not applicable to CDMA soft handoff. These
   considerations do not, however, apply to hard handoff which occurs in
   the core network after frame selection.

8.0  References

  [1] Perkins, C. (ed.), IP Mobility Support for IPv4, revised, draft-
        ietf-mobileip-rfc2002-bis.txt (work in progress), January, 2000.

  [2] Vakil, Faramak, et. al., Host Mobility Management Protocol:
        Extending SIP to 3G-IP Networks, draft-itsumo-hmmp-00.txt (work
        in progress), October, 1999.

  [3] E. Wedlund and H. Schulzrinne. Mobility Support Using SIP.  Second
        ACM/IEEE International Conference on Wireless and Mobile
        Multimedia (WoWMoM'99). Seattle, Washington. Aug. 1999.

  [4] O'Neill, A. and Corson, S., Edge Mobility Architecture, draft-
        oneill-ema-00.txt (work in progress), October, 1999.

  [5] Campbell, A., et. al., Cellular IP, draft-ietf-mobileip-



Kempf, McCann, Roberts    expires October 2000                 [Page 10]

INTERNET DRAFT                                                April 2000


        cellularip-00.txt, January, 2000.

  [6] Kempf, J. and Calhoun, P., Foreign Agent Assisted Hand-off,
        draft-calhoun-mobileip-proactive-fa-00.txt (work in progress),
        January, 2000.

 [7]  Ramjee, R., et. al., IP micro-mobility support using HAWAII,
        Internet Draft (work in progress), June 1999.

9.0  Authors' Addresses

   Questions about this memo can be directed to:

      James Kempf
      Network and Security Research Center, Sun Labs
      Sun Microsystems, Inc.
      901 San Antonio Rd., UMPK15-214
      Palo Alto, CA, 94303
      USA

       Phone: +1 650 786 5890
         Fax: +1 650 786 6445
      E-Mail: james.kempf@sun.com

      Peter McCann
      Bell Laboratories
      263 Shuman Boulevard
      Room 2Z-305
      P.O. Box 3050
      Naperville, IL, 60566
      USA

       Phone: +1 630 713 9359
         Fax: +1 630 713 4982
      E-Mail: mccap@research.bell-labs.com

      Philip Roberts
      Motorola, Inc.
      1501 W. Shure Dr.
      Arlington Heights, IL, 60015

      Phone:  +1 847 632 3148
      E-Mail: qa3445@email.mot.com




9.0  Full Copyright Statement



Kempf, McCann, Roberts    expires October 2000                 [Page 11]

INTERNET DRAFT                                                April 2000


   Copyright (C) The Internet Society (2000).  All Rights Reserved.

   This document and translations of it may be copied  and  furnished
   to others,  and  derivative works that comment on or otherwise
   explain it or assist in its implementation may be prepared, copied,
   published and distributed,  in  whole  or  in part, without
   restriction of any kind, provided that the  above  copyright  notice
   and  this  paragraph  are included on all such copies and derivative
   works.  However, this docu- ment itself may not be modified in any
   way, such as  by  removing  the copyright notice or references to the
   Internet Society or other Inter- net organizations, except as needed
   for  the  purpose  of  developing Internet standards in which case
   the procedures for copyrights defined in the Internet Standards
   process must be followed, or as required  to translate it into
   languages other than   English.  The limited permis- sions granted
   above are perpetual and  will  not  be  revoked  by  the Internet
   Society or its successors or assigns.  This document and the
   information contained herein is provided on an "AS IS" basis  and
   THE INTERNET SOCIETY AND THE INTERNET ENGINEERING TASK FORCE
   DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT
   LIMITED TO ANY  WAR- RANTY  THAT  THE  USE  OF THE INFORMATION HEREIN
   WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
   MERCHANTABILITY OR FITNESS  FOR  A PARTICULAR PURPOSE."




























Kempf, McCann, Roberts    expires October 2000                 [Page 12]


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From sip-admin@lists.bell-labs.com  Wed May  3 11:24:55 2000
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Date: Wed, 03 May 2000 11:24:45 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Dear IANA:

ftp://ftp.isi.edu/in-notes/iana/assignments/url-schemes is missing the
sip: URL scheme defined in RFC 2543, a Proposed Standard. What is the
process of having it added to the list?

Thank you.

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


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From sip-admin@lists.bell-labs.com  Wed May  3 12:53:40 2000
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From: "Tricia Wang" <twang@netrue.com>
To: "sip" <sip@lists.bell-labs.com>
Date: Wed, 3 May 2000 09:55:05 -0700
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Subject: [SIP] Accounting and Biling in SIP
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Hello,
   Can anyone help me understand how SIP deal with accounting, billing and
CDR (call detail record).
My understanding is that, proxy server or redirect server will save a log
file each time a call terminates.
And we can use the log files save to actually generates CDR.  Is this
correct?


Thanks

Tricia





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From sip-admin@lists.bell-labs.com  Wed May  3 12:59:41 2000
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To: Tricia Wang <twang@netrue.com>
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Subject: Re: [SIP] Accounting and Biling in SIP
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See the SIP FAQ at http://www.cs.columbia.edu/sip/

Tricia Wang wrote:
> 
> Hello,
>    Can anyone help me understand how SIP deal with accounting, billing and
> CDR (call detail record).
> My understanding is that, proxy server or redirect server will save a log
> file each time a call terminates.
> And we can use the log files save to actually generates CDR.  Is this
> correct?
> 

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


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> I'm not sure I see what is gained by adding a user portion. What is the
> semantic of having a user portion in the request URI? This still doesn't
> change how a proxy looks up registrations when an INVITE request is
> received (still based on request URI), which is what Dean is asking for.
> 

But it could open the possibility of changing how the REGISTER request gets routed/handled
based on the user portion of the Request-URI. (The routing I'm speaking of is outside
the realm of SIP, but shouldn't be precluded by the SIP protocol). In other words,
many SIP clients will be configurable as to where to send it's periodic REGISTER messages.
Allowing this configuration to be a full SIP URL including a user should make no
difference to the client, but may make a difference to the registrar. Probably 90% of
the registrars that get deployed won't use the user portion, but there's no reason
why those remaining 10% couldn't. Treating the URL as opaque (from the client perspective)
is a useful service primitive.

> -Jonathan R. 
> -- 
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com

--
Sean Olson <sean.olson@ericsson.com>



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HI James & co,  I have few questions after my first reading of the draft
Page 7:
             When you talk about RAN-RAN interconnection (closley coupled way),
what is your assumption? Does each neighboring  RAN (RAN-GW) belong to a
different IP subnet? or these RAN-GWs belong to the same subnet but are just
distributed along.

Page 6:

               Client listening to multiple base stations (3 or 6) at the same
time: Do you assume multiple interfaces on the client or just one?
In cases where the neighboring BSs for a mobile client are part of RANs
belonging to different subnets ( and controlled by different RAN-GW), how does
the RAN-GWs interact with each other to take care of soft-handoff, probably some
sort of IP-mobility solution like IP multicast can be helpful here.

Thanks
Ashutosh





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Date: Wed, 03 May 2000 12:13:39 -0700
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [SIP] Problem with the syntax of URI extension parameters
Cc: Igor Slepchin <islepchin@dynamicsoft.com>,
        Jonathan Lennox <lennox@ober.cs.columbia.edu>,
        sip@lists.research.bell-labs.com
In-Reply-To: <390F9E8E.10FAEAB1@cs.columbia.edu>
References: <200005022203.SAA34631@conrail.cs.columbia.edu>
 <390F9700.F3FA43D8@dynamicsoft.com>
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So in which contexts do i need to escape the <"> in this parameter?
      ?Request-Contact=*;mobility="mobile"

(obviously when the sip:URL is encoded in HTML)

thanks,
-rohan

At 08:35 PM 5/2/00 , Henning Schulzrinne wrote:
>Igor Slepchin wrote:
> >
> > You would probably want to exclude "?" and "," from param-reserved: "?"
> > -  to eliminate the ambiguity with "?" that separates parameters from
>
>Done.
>
> > headers, "," - because URLs are allowed in certain headers and allowing
> > commas in SIP URLs will make parsing those headers sometime impossible.
>
>As far as I know, all headers where URLs are allowed mandate the use of
><> for URLs with commas or other "bad" characters. Any we missed?
>(Probably still a good idea to make it reserved; the pain is small
>compared to the confusion avoided.)
>
> >
> > Another observation: to be as backward compatible as possible the
> > characters allowed in param-reserved should be a subset of those allowed
> > in token. The intersection of reserved from RFC2396 and token from
> > RFC2543 leaves us with the following
> >
> > param-reserved = "+"
>
>Anybody object to this most restrictive coding? The only slight problem
>with being restrictive is that it makes hand-coding HTML pages with SIP
>URLs more painful. On the other hand, we haven't really defined any
>parameters likely to see $ or @. However, removing : doesn't work well,
>since we allow it in IPv6 numerical addresses in the maddr parameter.
>Would seem strange to force escaping here.
>
> >
> > Also, note that unreserved in RFC2396 allows "(" and ")", while token
> > does not. On the other hand, I don't think that many implementations
> > will get broken if those extra characters are allowed.
> >
>
>
>--
>Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
>
>
>_______________________________________________
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Date: Wed, 03 May 2000 14:37:41 -0500
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: [SIP] Second try : Questions about REGISTER
In-reply-to: <200005031732.MAA18556@b04a45.exu.ericsson.se>
To: Sean Olson <eussean@exu.ericsson.se>, jdrosen@dynamicsoft.com
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Jonathan said:
> > I'm not sure I see what is gained by adding a user portion. What is the
> > semantic of having a user portion in the request URI? This still doesn't
> > change how a proxy looks up registrations when an INVITE request is
> > received (still based on request URI), which is what Dean is asking for.
> >

and Sean said:
> But it could open the possibility of changing how the REGISTER
> request gets routed/handled
> based on the user portion of the Request-URI. (The routing I'm
> speaking of is outside
> the realm of SIP, but shouldn't be precluded by the SIP
> protocol). In other words,
> many SIP clients will be configurable as to where to send it's
> periodic REGISTER messages.
> Allowing this configuration to be a full SIP URL including a user
> should make no
> difference to the client, but may make a difference to the
> registrar. Probably 90% of
> the registrars that get deployed won't use the user portion, but
> there's no reason
> why those remaining 10% couldn't. Treating the URL as opaque
> (from the client perspective)
> is a useful service primitive.

Does an INVITE have to be delivered only to the node indicated in the
right-hand-side of the request-URI? No, because we require transitivity
support in a proxy -- that is, we can ask a proxy to resolve something that
we know is not local to that proxy. That's what makes it a proxy, from my
point of view. Similarly, the To: and request-URI can differ due to request
transformations that have occurred in the proxy chain.

This is different from REGISTER. From 4.2.6 "The Request-URI names the
destination of the         registration request, i.e., the domain of the
registrar". Why? So a register can be proxied -- that is, a proxy/registrar
can decide whether the REGISTER was meant for it, and if not resolve the
next hop towards a registrar using the request-URI. Or so that a multidomain
registrar can determine which domain the registration should be valid in. I
argue more granularity is needed.

So, what would be the semantic of (if it were allowed)
	REGISTER dwillis@softarmor.com
	To: dean.willis@wcom.com
	Contact:@63.64.250.84

What I want this to mean is "Within the registrar "softarmor.com" there is
an identity "dwillis". Calls directed to this identity which have a "To:"
value of dean.willis@wcom.com should be diverted to @63.64.250.85". The
registration being changed is the one for the identity
"dwillis@softarmor.com", NOT the identity "dean.willis@wcom.com".

This would further imply:

1) Mimicing today's behavior would require moving to To: value to the
request-URI. That is:
	REGISTER dwillis@softarmor.com
	Contact:@63.64.250.84
   would, in the absence of other registrations and rules, forward any
INVITE with a request-URI of dwillis@softarmor.com to the UAS @63.64.250.84

2) Traditional behavior would occur with a To: which matches the request-URI
iff the proxy/registrar does not offer To: field matching.

The other question is, what would the implications be for authentication?
Essentially, we would be saying that the registration target is the body of
the request-URI, not the body of the To: I find this far more consistent
with the semantics of INVITE and OPTIONS than the current approach is.

--
Dean



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To: Ge Yu <engp9374@nus.edu.sg>
Cc: Shail Bhatnagar <shbhatna@cisco.com>,
        "'Alexandru Gavrilescu'" <alexang@Exchange.Microsoft.com>,
        SIP IETF <sip@lists.research.bell-labs.com>
Subject: Re: [SIP] Second try : Questions about REGISTER
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Ge Yu wrote:
> 
> Hi, friends,
> 
> As I know through 16.1 of RFC 2543, the registration process is as follows:
> 
> 1. Watson firstly sends the REGISTER request to the server at bell-tel.com,
> with Contact: Watson@example.com. That is, he hasn't contacted with the
> server at example.com ahead of REGISTER.
>     So what happens if another Watson@example.com has been registered in
> example.com domain?

Watson wouldn't send the registration with Contact: watson@example.com
unless he was sure he owned that name at example.com. The assumption,
stated in 16.1, is that watson has visiting rights at example.com, and
so he knows that watson@example.com is his own address.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Subject: Re: [SIP] Second try : Questions about REGISTER
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Dean Willis wrote:
> 
> Jonathan said:
> > > I'm not sure I see what is gained by adding a user portion. What is the
> > > semantic of having a user portion in the request URI? This still doesn't
> > > change how a proxy looks up registrations when an INVITE request is
> > > received (still based on request URI), which is what Dean is asking for.
> > >
> 
> and Sean said:
> > But it could open the possibility of changing how the REGISTER
> > request gets routed/handled
> > based on the user portion of the Request-URI. (The routing I'm
> > speaking of is outside
> > the realm of SIP, but shouldn't be precluded by the SIP
> > protocol). In other words,
> > many SIP clients will be configurable as to where to send it's
> > periodic REGISTER messages.
> > Allowing this configuration to be a full SIP URL including a user
> > should make no
> > difference to the client, but may make a difference to the
> > registrar. Probably 90% of
> > the registrars that get deployed won't use the user portion, but
> > there's no reason
> > why those remaining 10% couldn't. Treating the URL as opaque
> > (from the client perspective)
> > is a useful service primitive.

> So, what would be the semantic of (if it were allowed)
>         REGISTER dwillis@softarmor.com
>         To: dean.willis@wcom.com
>         Contact:@63.64.250.84
> 
> What I want this to mean is "Within the registrar "softarmor.com" there is
> an identity "dwillis". Calls directed to this identity which have a "To:"
> value of dean.willis@wcom.com should be diverted to @63.64.250.85". The
> registration being changed is the one for the identity
> "dwillis@softarmor.com", NOT the identity "dean.willis@wcom.com".

This is not the semantics of REGISTER as they are today; I would be
unfortable changing the semantics like this because a user portion has
been added to the REGISTER request URI. A proxy would have very
different rules for routing depending on how the registration looked.

If you want non-standard routing, have the clients or administrators
upload CPL or CGI scripts to do these things.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From: Francois-Xavier Guitton <Francois-Xavier.Guitton@ms.alcatel.fr>
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All,

When I read RFC2543, Annex B - Usage of SDP p 136, I am a little bit
confused. Could you answer the following questions :

1/ RFC 2327 (SDP) defines 3 attributes a=recvonly, a=sendrcv and
a=sendonly. But, in SIP, do we consider a media is by default
bi-directional ? The example p 137 seems to tell that, but does someone
know where that is written ?

2/ Is it possible for a user agent to add new media in the SDP part of
the answer to a received INVITE?

For example A INVITE B with  media:
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000

Does it make sense B answers 200 OK with:
m=audio 4910 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 5300 RTP/AVP 32
a=rtpmap:32 MPV/90000

The underlying question being: What type of media are listed in the SDP
part : the media supported by the User Agent ? or the media the Human
user wishes to use in the SIP session ? In that case, A could wish an
audio session but B would prefer an audio-video session. If we suppose A
supports also video capabilities, what will answer A : ACK with audio +
video media description ??? Or is a new INVITE from A to B required ?

3/ Where is defined the list of all the types of rtpmap (e.g.: PCMU,
H261, MPV ...) supported by SIP in the SDP part ?

Regards,

Fxg

PS: What is the status of draft-ietf-sip-rfc2543bis ? Is it now the
reference instead of RFC 2543 ?

./.

--
Francois-Xavier Guitton (SWD/ULC)
mailto:Francois-Xavier.Guitton@ms.alcatel.fr
Tel: +33 (0)1 69 63 47 52
Fax: +33 (0)1 69 63 17 89



--------------E1B13077538FF7B5006113C7
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
All,
<p>When I read RFC2543, Annex B - Usage of SDP p 136, I am a little bit
confused. Could you answer the following questions :
<p>1/ RFC 2327 (SDP) defines 3 attributes a=recvonly, a=sendrcv and a=sendonly.
But, in SIP, do we consider a media is by default bi-directional ? The
example p 137 seems to tell that, but does someone know where that is written
?
<p>2/ Is it possible for a user agent to add new media in the SDP part
of the answer to a received INVITE?
<p>For example A INVITE B with&nbsp; media:
<br>m=audio 49170 RTP/AVP 0
<br>a=rtpmap:0 PCMU/8000
<p>Does it make sense B answers 200 OK with:
<br>m=audio 4910 RTP/AVP 0
<br>a=rtpmap:0 PCMU/8000
<br>m=video 5300 RTP/AVP 32
<br>a=rtpmap:32 MPV/90000
<p>The underlying question being: What type of media are listed in the
SDP part : the media supported by the User Agent ? or the media the Human
user wishes to use in the SIP session ? In that case, A could wish an audio
session but B would prefer an audio-video session. If we suppose A supports
also video capabilities, what will answer A : ACK with audio + video media
description ??? Or is a new INVITE from A to B required ?
<p>3/ Where is defined the list of all the types of rtpmap (e.g.: PCMU,
H261, MPV ...) supported by SIP in the SDP part ?
<p>Regards,
<p>Fxg
<p>PS: What is the status of draft-ietf-sip-rfc2543bis ? Is it now the
reference instead of RFC 2543 ?
<p>./.
<pre>--&nbsp;
Francois-Xavier Guitton (SWD/ULC)
<A HREF="mailto:Francois-Xavier.Guitton@ms.alcatel.fr">mailto:Francois-Xavier.Guitton@ms.alcatel.fr</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Tel: +33 (0)1 69 63 47 52&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Fax: +33 (0)1 69 63 17 89</pre>
&nbsp;</html>

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Subject: FW: [SIP] questions for clarification on SDP negociation
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Francois, see comments below marked [Itamar Gilad].
 
  Itamar

-----Original Message-----
From: Francois-Xavier Guitton [mailto:Francois-Xavier.Guitton@ms.alcatel.fr]
Sent: Thu, May 04, 2000 9:17 AM
To: SIP
Subject: [SIP] questions for clarification on SDP negociation


All, 

When I read RFC2543, Annex B - Usage of SDP p 136, I am a little bit
confused. Could you answer the following questions : 


1/ RFC 2327 (SDP) defines 3 attributes a=recvonly, a=sendrcv and a=sendonly.
But, in SIP, do we consider a media is by default bi-directional ? The
example p 137 seems to tell that, but does someone know where that is
written ? 


[Itamar Gilad] I'm not sure where it's written, but the general idea as I
understand it is that by default the sender of the INVITE informs of the
media it is willing to send AND receive.  If you wish to specify asymmetric
send/receive capabilities you need to set up two m= lines, one with
a=sendonly and one with a=recvonly. 



2/ Is it possible for a user agent to add new media in the SDP part of the
answer to a received INVITE? 


For example A INVITE B with  media: 
m=audio 49170 RTP/AVP 0 
a=rtpmap:0 PCMU/8000 


Does it make sense B answers 200 OK with: 
m=audio 4910 RTP/AVP 0 
a=rtpmap:0 PCMU/8000 
m=video 5300 RTP/AVP 32 
a=rtpmap:32 MPV/90000 


The underlying question being: What type of media are listed in the SDP part
: the media supported by the User Agent ? or the media the Human user wishes
to use in the SIP session ? In that case, A could wish an audio session but
B would prefer an audio-video session. If we suppose A supports also video
capabilities, what will answer A : ACK with audio + video media description
??? Or is a new INVITE from A to B required ? 


[Itamar Gilad] B can't add new media which A did not specify in the original
INVITE. It can either accept a subset or whole of the the media streams and
codecs suggested by A or decline the invitation. Once a session has been
established either side can attempt adding new media streams by sending a
re-INVITE with an appropriate SDP body (see B.5 in RFC2543-bis).  

3/ Where is defined the list of all the types of rtpmap (e.g.: PCMU, H261,
MPV ...) supported by SIP in the SDP part ? 


[Itamar Gilad] As far as I know SIP does not set any limit on the types of
media describable in the SDP body.  See also RFC1889 and RFC1890.  

Regards, 


Fxg 


PS: What is the status of draft-ietf-sip-rfc2543bis ? Is it now the
reference instead of RFC 2543 ? 


./. 

-- 

Francois-Xavier Guitton (SWD/ULC)

mailto:Francois-Xavier.Guitton@ms.alcatel.fr
<mailto:Francois-Xavier.Guitton@ms.alcatel.fr>       

Tel: +33 (0)1 69 63 47 52                    

Fax: +33 (0)1 69 63 17 89
  


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<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1">


<META content="MSHTML 5.00.2314.1000" name=GENERATOR></HEAD>
<BODY>
<DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
size=2><BR></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=628393608-04052000>Francois, see comments below marked [Itamar 
Gilad].</SPAN></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=628393608-04052000>&nbsp; 
Itamar</SPAN></FONT></DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Francois-Xavier Guitton 
  [mailto:Francois-Xavier.Guitton@ms.alcatel.fr]<BR><B>Sent:</B> Thu, May 04, 
  2000 9:17 AM<BR><B>To:</B> SIP<BR><B>Subject:</B> [SIP] questions for 
  clarification on SDP negociation<BR><BR></DIV></FONT>All, 
  <P>When I read RFC2543, Annex B - Usage of SDP p 136, I am a little bit 
  confused. Could you answer the following questions : 
  <P>1/ RFC 2327 (SDP) defines 3 attributes a=recvonly, a=sendrcv and 
  a=sendonly. But, in SIP, do we consider a media is by default bi-directional ? 
  The example p 137 seems to tell that, but does someone know where that is 
  written ? 
  <P><SPAN class=628393608-04052000><FONT color=#0000ff face=Arial 
  size=2>[Itamar Gilad]&nbsp;I'm not sure where it's written, but the general 
  idea as I understand it is that by default the sender of the INVITE informs of 
  the media it is willing to send AND receive.&nbsp; If you&nbsp;wish to specify 
  asymmetric send/receive capabilities you need to set up two m= lines, one with 
  a=sendonly and one with a=recvonly. </FONT></SPAN>
  <P><SPAN class=628393608-04052000></SPAN><BR>2/ Is it possible for a user 
  agent to add new media in the SDP part of the answer to a received INVITE? 
  <P>For example A INVITE B with&nbsp; media: <BR>m=audio 49170 RTP/AVP 0 
  <BR>a=rtpmap:0 PCMU/8000 
  <P>Does it make sense B answers 200 OK with: <BR>m=audio 4910 RTP/AVP 0 
  <BR>a=rtpmap:0 PCMU/8000 <BR>m=video 5300 RTP/AVP 32 <BR>a=rtpmap:32 MPV/90000 

  <P>The underlying question being: What type of media are listed in the SDP 
  part : the media supported by the User Agent ? or the media the Human user 
  wishes to use in the SIP session ? In that case, A could wish an audio session 
  but B would prefer an audio-video session. If we suppose A supports also video 
  capabilities, what will answer A : ACK with audio + video media description 
  ??? Or is a new INVITE from A to B required ? <BR><FONT color=#0000ff 
  face=Arial size=2><SPAN class=628393608-04052000></SPAN></FONT></P>
  <P><FONT color=#0000ff face=Arial size=2><SPAN 
  class=628393608-04052000>[Itamar Gilad]</SPAN></FONT><FONT color=#0000ff 
  face=Arial size=2><SPAN class=628393608-04052000>&nbsp;B can't add new media 
  which A did not specify in the original INVITE. It&nbsp;can either accept a 
  subset or whole of the the media streams&nbsp;and codecs suggested by A or 
  decline the invitation.&nbsp;Once a session has been established&nbsp;either 
  side can&nbsp;attempt adding new media streams by sending a 
  re-INVITE&nbsp;with an appropriate SDP body (see B.5 in RFC2543-bis).&nbsp; 
  </SPAN></FONT></P>
  <P>3/ Where is defined the list of all the types of rtpmap (e.g.: PCMU, H261, 
  MPV ...) supported by SIP in the SDP part ? <BR><FONT color=#0000ff face=Arial 
  size=2><SPAN class=628393608-04052000></SPAN></FONT></P>
  <P><FONT color=#0000ff face=Arial size=2><SPAN 
  class=628393608-04052000>[Itamar Gilad]&nbsp;As far as I know SIP does not set 
  any limit on the types of media&nbsp;describable in the SDP 
  body.&nbsp;&nbsp;See&nbsp;also RFC1889 and 
  RFC1890.&nbsp;&nbsp;</SPAN></FONT></P>
  <P>Regards, 
  <P>Fxg 
  <P>PS: What is the status of draft-ietf-sip-rfc2543bis ? Is it now the 
  reference instead of RFC 2543 ? 
  <P>./. <PRE>--&nbsp;
Francois-Xavier Guitton (SWD/ULC)
<A href="mailto:Francois-Xavier.Guitton@ms.alcatel.fr">mailto:Francois-Xavier.Guitton@ms.alcatel.fr</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Tel: +33 (0)1 69 63 47 52&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Fax: +33 (0)1 69 63 17 89</PRE>&nbsp; </BLOCKQUOTE></BODY></HTML>

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	Title		: IP Mobility and the CDMA Radio Access Network:
                          Applicability Statement for Soft Handoff
	Author(s)	: J. Kempf,  P. McCann, P. Roberts 
	Filename	: draft-kempf-cdma-appl-00.txt
	Pages		: 12
	Date		: 03-May-00
	
Recently, there have been a variety of proposals submitted to the
Mobile IP Working Group and to other IETF working groups for IP
mobility solutions that seek to enhance or replace mobile IP. These
proposals, often characterized as micromobility or fast handoff, are
addressed primarily at the perceived need of multimedia sessions such
as video or voice over IP for faster handoff between radio base
stations, and are primarily directed at real time multimedia traffic
in 3rd generation cellular access networks. In this paper, we discuss
the design of CDMA radio access networks (RANs) and the applicability
of IP mobility to soft handoff in a CDMA RAN. We attempt to show that
given current IP routing algorithms and the constraints on a CDMA
RAN, IP mobility solutions have little, if any, role to play in
handoff within the RAN. In contrast, an IP mobility solution is
likely to play a big role in fast handoff between RANs, also called
hard handoff. While future developments in IP networking may change
this situation, IP mobility in CDMA networks currently seems to apply
only when the mobile node roams between RANs rather than between base
stations within a RAN.

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From: James Kempf <James.Kempf@eng.sun.com>
Reply-To: James Kempf <James.Kempf@eng.sun.com>
Subject: Re: [SIP] Applicability of IP Mobility to CDMA Soft Handoff
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>HI James & co,  I have few questions after my first reading of the draft
>Page 7:
>             When you talk about RAN-RAN interconnection (closley coupled way),
>what is your assumption? Does each neighboring  RAN (RAN-GW) belong to a
>different IP subnet? or these RAN-GWs belong to the same subnet but are just
>distributed along.
>

Whether or not they belong to a different subnet is not important from the
point of view of a moving terminal, since the terminal's IP packets
are not routed through the RAN based on the terminal's IP address.
The termimal's packets are processed into radio frames, in which
the IP packets are encoded. If the RAN is IP based (it may not be and, in fact, 
in current 2G and 3G designs is not), then the links are typically point to 
point or at most one hop, due to the stringent timing constraints. The
radio frames in an IP-based RAN would run on top of IP (maybe UDP) directly, and
the terminal's IP packets would not be identifiable in the RAN as 
such, except by performing the same processing the terminal's CDMA
hardware performs to decode the radio frames.

>Page 6:
>
>               Client listening to multiple base stations (3 or 6) at the same
>time: Do you assume multiple interfaces on the client or just one?
>In cases where the neighboring BSs for a mobile client are part of RANs
>belonging to different subnets ( and controlled by different RAN-GW), how does
>the RAN-GWs interact with each other to take care of soft-handoff, probably 
some
>sort of IP-mobility solution like IP multicast can be helpful here.
>

The client has only one "interface" in the IP sense. A mobile CDMA
phone uses a signal processing technique to combine the incoming
signals to produce a single packet. This process proceeds at
the physical layer, i.e. L1, and therefore the upper layers never
see more than a single packet as was incoming to the frame selector/
distributor from the corresponding node. So the phone can have only a single IP 
address for the CDMA connection, even though at the physical layer there are 
multiple signals coming in with the same (modulo errors) octet stream.

This characteristic is the single largest deterrent to using IP
mobility in CDMA RANs.

		jak



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From: "Hans-henrik Veisner (DXD)" <Hans-henrik.W.Veisner@dxd.ericsson.se>
To: sip@lists.research.bell-labs.com
Subject: RE: [SIP] Content Lenght in responses
Date: Thu, 4 May 2000 17:30:01 +0200 
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Hi!

What about the case where a UAC sends an INVITE and the UAS returns a 18x response and closes the connection to indicate the end of the message. Shouldn't the UAC interpret this as a 500 response as described in section 10.3 in the rfc2543bis draft?

Hans-Henrik Veisner,
ANS IP Convergence
Ericsson Diax


-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: 3. maj 2000 06:33
To: Natarajan Kalpathy
Cc: sip@lists.research.bell-labs.com
Subject: Re: [SIP] Content Lenght in responses




Natarajan Kalpathy wrote:
> 
> Why is the Content Length field not mandatory in cases of SIP responses ??. Wont
> it simplify the message processing in case of TCP. The current rfc2543bis draft
> says
> 
> "If a response does not contain a Content-Length, the client assumes that it
> encompasses the remainder
> of the UDP packet or the data until the TCP connection is closed, as applicable.
> "
> 
> Please clarify ....

I believe this is the result of legacy HTTP, where Content-Length is not
mandatory in responses. I also believe this is because CGI scripts that
generate dynamic content wouldn't know the Content Length ahead of time,
so just sending data, and then closing the connection, was the easiest
way to determine message length. I can certainly envision placing of
dynamic content in SIP responses, so this may prove useful for us as
well.

Processing of responses without content length is pretty simple anyway.
It just means an additional counter that counts actual bytes read from
the response. When the connection closes, this value is your content
length.

-Jonathan R.



-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Thu May  4 16:47:40 2000
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From: "Eber Mello" <eber@ss8networks.com>
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Subject: [SIP] Registrar providing location services
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Hi,

Please, I have a few questions regarding the behavior of a SIP
registrar providing location services vis-a-vis the
action (proxy or redirect) present on the REGISTER message.

1) Should a proxy- *and* redirect capable SIP server chooses
either to proxy or redirect a request based on the policy of the
user addressed on the request, expressed by the user's REGISTER
message? In other words, the location services provided by the
registrar would return how - proxy or redirect, the user wants
to be reached as well as a contact list?

2) Can a user register contacts that are actually
well known SIP URLs of other users, as below?

userA@domainX registers with action=PROXY, contact1=userB@domainY
and contact2 = userC@domainZ.

3) Consider that userB and userC are also registered with the same
registrar,
as follows:

userB@domainY registers with action=REDIRECT, contact=addressK@domainM
and userC@domainZ registers with action=PROXY, contact=addressL@domainN

=> lets suppose that addressK@domainM and addressL@domainN are not
locally registered

When queried about the location of 'userA', should the location server
recurse on the contacts of userA whenever it sees a SIP URL of a user
also registered locally to it?

4) If the answer to the previous questions are YES, than the location server
would return REDIRECT to addressK@domainM and PROXY to addressL@domainN -
what is
a scenario that the spec explicitly tries to avoid by enforcing that
all contacts on a REGISTER must have the same 'action'.

Thanks,

Eber Mello
SS8 Networks Canada.
www.ss8networks.com
phone: 613-5921418



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From sip-admin@lists.bell-labs.com  Thu May  4 18:03:27 2000
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Date: Thu, 04 May 2000 17:58:33 -0400
From: Igor Slepchin <islepchin@dynamicsoft.com>
Organization: dynamicsoft Inc.
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Subject: Re: [SIP] Registrar providing location services
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Eber Mello wrote:
> 
> 1) Should a proxy- *and* redirect capable SIP server chooses
> either to proxy or redirect a request based on the policy of the
> user addressed on the request, expressed by the user's REGISTER
> message? 

That's up to you. You can either honor user preferences or overwrite
them.
 
> 2) Can a user register contacts that are actually
> well known SIP URLs of other users, as below?

Sure.
 
> 
> When queried about the location of 'userA', should the location server
> recurse on the contacts of userA whenever it sees a SIP URL of a user
> also registered locally to it?

If domainM and domainN are served by the same proxy, it's a nice idea to
do the recursive lookup. Nothing catastrophic happens if you don't - the
proxy will just have to do the same lookup when it gets the request it
just proxied to itself.
 
> 4) If the answer to the previous questions are YES, than the location server
> would return REDIRECT to addressK@domainM and PROXY to addressL@domainN -
> what is
> a scenario that the spec explicitly tries to avoid by enforcing that
> all contacts on a REGISTER must have the same 'action'.

I guess the question is about what to do when the recursed Contact has
the action different from that in the first Contact. Well, that's up to
the proxy: it can overwrite that action or, if this turns out to be the
last Contact to try, it can overwrite the action from the first Contact.

---
Igor Slepchin


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draft-ietf-sip-rfc2543bis-00 refers to Content-Language header
in 6.46, but it does not seem to formally define this header.
Am I missing something?

Clark


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Date: Thu, 04 May 2000 18:51:25 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Subject: [SIP] [Fwd: SIP URL scheme]
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This is a multi-part message in MIME format.
--------------B64CD6FCAE296882F4450B90
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Thus, whoever was dealing with the Mozilla folks can now point to the
document

http://www.isi.edu/in-notes/iana/assignments/url-schemes

to prove legitimacy.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
--------------B64CD6FCAE296882F4450B90
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Reply-To: <iana@iana.org>
From: "IANA" <iana@ISI.EDU>
To: "'Henning Schulzrinne'" <schulzrinne@cs.columbia.edu>
Subject: RE: SIP URL scheme
Date: Thu, 4 May 2000 15:18:37 -0700
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Dear Henning,

Thank you for notifying the IANA.  This has been added.

Best regards,

Michelle Schipper, Administrator
IANA

-----Original Message-----
From: hgs@cs.columbia.edu [mailto:hgs@cs.columbia.edu]On Behalf Of
Henning Schulzrinne
Sent: Wednesday, May 03, 2000 8:25 AM
To: iana@ISI.EDU
Cc: sip@lists.bell-labs.com; Mathieu Gervais (LMC)
Subject: SIP URL scheme


Dear IANA:

ftp://ftp.isi.edu/in-notes/iana/assignments/url-schemes is missing the
sip: URL scheme defined in RFC 2543, a Proposed Standard. What is the
process of having it added to the list?

Thank you.

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



--------------B64CD6FCAE296882F4450B90--



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"Hans-henrik Veisner (DXD)" wrote:
> 
> Hi!
> 
> What about the case where a UAC sends an INVITE and the UAS returns a 18x response and closes the connection to indicate the end of the message. Shouldn't the UAC interpret this as a 500 response as described in section 10.3 in the rfc2543bis draft?

Yes. If you still have more responses to send after the 18x, include a
Content-Length rather than closing the connection.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Thu May  4 22:55:14 2000
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To: Igor Slepchin <islepchin@dynamicsoft.com>
Cc: Eber Mello <eber@ss8networks.com>, sip@lists.bell-labs.com
Subject: Re: [SIP] Registrar providing location services
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Some additional comments:

Igor Slepchin wrote:
> 
> > 4) If the answer to the previous questions are YES, than the location server
> > would return REDIRECT to addressK@domainM and PROXY to addressL@domainN -
> > what is
> > a scenario that the spec explicitly tries to avoid by enforcing that
> > all contacts on a REGISTER must have the same 'action'.
> 
> I guess the question is about what to do when the recursed Contact has
> the action different from that in the first Contact. Well, that's up to
> the proxy: it can overwrite that action or, if this turns out to be the
> last Contact to try, it can overwrite the action from the first Contact.

It should generally do whatever would happen if the domains were all
served by proxies. So, given the description of the registrations:

> userA@domainX registers with action=PROXY, contact1=userB@domainY
> and contact2 = userC@domainZ.
> userB@domainY registers with action=REDIRECT, contact=addressK@domainM
> and userC@domainZ registers with action=PROXY, contact=addressL@domainN

 the following would happen:

1. an INVITE arrives at proxy1 for userA@domainX. Based on its
registrations, it proxies this to userB@domainY and userC@domainZ

2. one invite goes to domainY and one to domainZ. domainY looks up
userB, and sends a 300 redirect with Contact addressk@domainM. domainZ
looks up userC, and proxies to addressL@domainN.

3. the 300 redirect arrives at the first proxy, which proxies to
addressk@domainM. The proxy for domainN receives the INVITE for
addressL@domainN, and does whatever its configured to do.


Now, should any of these domains be served by the same proxy, the
behavior should be exactly the same as if they were not served by the
same proxy. So, if domainX, domainY, domainZ are all served by one
proxy, that proxy would have ended up proxying the received request to
addressk@domainM and addressL@domainN.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Thu May  4 23:00:08 2000
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This is also present in rfc2543, and I believe it is an error resulting
from cut and paste from HTTP, which contains this paragraph verbatim.
There is no Content-Language header in SIP. This should be stricken from
rfc2543bis. Thanks for pointing it out.

-Jonathan R.

"G. Clark Brown" wrote:
> 
> draft-ietf-sip-rfc2543bis-00 refers to Content-Language header
> in 6.46, but it does not seem to formally define this header.
> Am I missing something?
> 
> Clark
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Jonathan Rosenberg wrote:
> 
> This is also present in rfc2543, and I believe it is an error resulting
> from cut and paste from HTTP, which contains this paragraph verbatim.
> There is no Content-Language header in SIP. This should be stricken from
> rfc2543bis. Thanks for pointing it out.

Hmm. If we allow HTML or other objects in the SIP response, maybe
Content-Language doesn't seem all that far-fetched. At least that way, I
can automatically invoke the babelfish translator when I get the Spanish
web-IVR menu :-)



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


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> Alexandru Gavrilescu wrote:
> 
> I have a question regarding Jonathan's answer to the Shail's third
> question.
> Do we assume that tawatson has an account created for himself at
> example.com or he is just a visitor at example.com?

Whether he is visiting or not, ann account must be set up for him at
example.com. Watson can't just assume a registration to example.com will
succeed without knowing ahead of time what his address there is.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Dear all,

The RFC2543 is not very clear regarding the SDP content of a 606 - Not
Acceptable response. I think it should be useful to specify in the 606
response all the media capabilities supported by the callee user agent.
It will allow the caller to send directly a new INVITE with the right
SDP description, without using supplementary OPTIONS discovery message,
or INVITE with no SDP.

Can we precise that point in the next edition of the RFC 2543 ???

Regards,

Francois-Xavier Guitton (SWD/ULC)

mailto:Francois-Xavier.Guitton@ms.alcatel.fr
Tel: +33 (0)1 69 63 47 52
Fax: +33 (0)1 69 63 17 89




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Subject: [SIP] Clarification to Figure 13 Bis Draft
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Hi All,

I think we all agree that the Bis draft is infinitely more readable than the
plain text forerunner. A big improvement.

As a result of the improved readability, I require a clarification to Figure 13
in section 11, which documents a State Transition Diagram of server for
INVITE Method.

My confusion mainly centres around the Failure state and subsequent
transitions.

1) If the Failure state receives an ACK, a transition to Confirmed occurs.
This doesn't make sense to me as there is now no way to revert to Initial
state. I can see how this works for the Success state, as the BYE event,
as a result of a hangup, causes the transition to Initial state.

Perhaps the ACK event in the Failure state should cause a transition to
Initial state?

2) A transition from Failure state to Initial state is shown on the diagram, as
a dashed line. Unfortunately, this transition is unlabelled. I'm referring
to the left hand dashed line, not the BYE event which joins the Success
dashed line.

3) Does the diagram need to be modified in light of changes in section
4.2.5 CANCEL, which talks about the issuing of a 487 (Request Cancelled)?
Presumably the 487 response is the response to the original INVITE, and a
separate 200 is issued as a response to the CANCEL?

4) This leads me into more questions about Cancel in section 4.2.5.
The draft states ....

"A redirect or user agent server receiving a CANCEL request responds with a
status of 200 (OK) if the transaction exists and a status of 481 (Transaction Does
Not Exist) if not, but takes no further action. In particular, any existing calls is
unaffected"

What defines "transaction does not exist" from a user agent server perspective?
Is a transaction deemed to be non-existent if the user agent server has issued
a final response?

If so then the Failure and Success states in Figure 13, presumably should issue
481 (Transaction Does Exist) as a result of CANCEL request?

The user agent client is then left with two scenarios:

a) Call failed at user agent server after Cancel was sent, so ACK >= 300 response
b) Call succeeded at user agent server after Cancel was sent, so ACK the 200
response, and issue BYE to terminate call.

So many questions, I hope that you will be able to follow my thread. :-)

Thanks In Advance
Phil

--
Ubiquity Software Corporation           http://www.ubiquity.net


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Subject: [SIP] 2543 bis: Section H.1 URI Syntax/header syntax
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Regarding header syntax in the Open Issues section, it's suggested that
hname and hvalue can't be *uric because of the obvious ambiguities. One
suggestion is to just disallow  "&" and "=". Another possibility is to
just remove reserved as an alternative in the uric rule, since the text
states that "All URL reserved characters in the header names and values
must be escaped", i.e., there's no point in including reserved in the
BNF for uric, at least for this purpose.

It seems the general intent is to allow the specification of SIP headers
in SIP URL's. Since the message header field-name [6.6] is defined to be
a token, this suggests that it's sufficent that hname is a token as
well. Unfortunately, the generic definition of field-value isn't quite
so helpful:

field-value = *(field-content | LWS)
field-content = <the OCTETs making up the field-value and consisting
                 of either *TEXT-UTF8 or combinations of token,
                 separators and quoted-string>

So perhaps the first thing to do is to replace the ABNF prose rules for
field-content and TEXT-UTF8 with something more formal, e.g.,

field-content = quoted-string | separator | token | TEXT-UTF8char

where

TEXT-UTF8 = TEXT-UTF8char | LWS
TEXT-UTF8char = %x21-7e | %x80-ff

Ideally, we'd like to equate header values to field values, i.e.,

hvalue = field-value

but field-value is even more general than the current 1*uric used for
hvalue, so the problem gets worse. In the absence, of a more useful
definition of field-content, there seem to be two approaches:

1. Follow the current spec and escape every reserved character. My own
feeling is that this renders most things somewhat unreadable (if that
matters), e.g.,

sip:j.doe@big.com;maddr=239.255.255.1;ttl=15

becomes:

sip%3aj.doe%big.com%3bmaddr%3d239.255.255.1%3bttl%3d15

2. The other approach is to use an "envelope" syntax, like
quoted-string, i.e.,

hvalue = token | quoted-string

Now only quotes have to be escaped (as a quoted-pair), but these seem to
be relatively infrequent in most of the SIP examples I've seen. The rule
for converting an hvalue to a field-content :
a) if it's a token, it is the field-content
b) if it's a quoted-string, remove the outer quotes and convert each
contained quoted-pair to corresponding character.

Now if quoted-string is a problem for the reason cited earlier in a
similar discussion on parameter names and values (double quote marks are
excluded from URI's), use a single quoted string instead. Single quote
is allowed in URI's and I don't think it's used for any SIP syntax.
However, this only excludes them from the outer level syntax unless we
want to exclude quoted-strings from all of SIP overall. (Or maybe just
from any SIP contained in URL headers.)

In any case, approach 2. seems more attractive to me, but I'm not sure
what current practice is in this area.

Rick Workman
Nortel Networks
Ottawa


P.S. Minor point: In section G "Changes to Be Made", a bug is recorded
for the Call-ID header which proposes to change the field-content to
token. I agree with the intent of the proposal, but I don't think token
allows @. This may break a lot of existing implementations since the
current RFC insists on @ being present. Also note that the current spec
also allows @ in the local-id which may cause a bit of confusion, but I
suspect most implementations just treat Call-ID as a single character
sequence anyway.


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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<tt></tt>&nbsp;
<br><tt>Regarding header syntax in the Open Issues section, it's suggested
that hname and hvalue can't be *uric because of the obvious ambiguities.
One suggestion is to just disallow&nbsp; "&amp;" and "=". Another possibility
is to just remove reserved as an alternative in the uric rule, since the
text states that "All URL reserved characters in the header names and values
must be escaped", i.e., there's no point in including reserved in the BNF
for uric, at least for this purpose.</tt>
<p><tt>It seems the general intent is to allow the specification of SIP
headers in SIP URL's. Since the message header field-name [6.6] is defined
to be a token, this suggests that it's sufficent that hname is a token
as well. Unfortunately, the generic definition of field-value isn't quite
so helpful:</tt>
<p><tt>field-value = *(field-content | LWS)</tt>
<br><tt>field-content = &lt;the OCTETs making up the field-value and consisting</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
of either *TEXT-UTF8 or combinations of token,</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
separators and quoted-string></tt>
<p><tt>So perhaps the first thing to do is to replace the ABNF prose rules
for field-content and TEXT-UTF8 with something more formal, e.g.,</tt>
<p><tt>field-content = quoted-string | separator | token | TEXT-UTF8char</tt><tt></tt>
<p><tt>where</tt><tt></tt>
<p><tt>TEXT-UTF8 = TEXT-UTF8char | LWS</tt>
<br><tt>TEXT-UTF8char = %x21-7e | %x80-ff</tt>
<p><tt>Ideally, we'd like to equate header values to field values, i.e.,</tt>
<p><tt>hvalue = field-value</tt>
<p><tt>but field-value is even more general than the current 1*uric used
for hvalue, so the problem gets worse. In the absence, of a more useful
definition of field-content, there seem to be two approaches:</tt>
<p><tt>1. Follow the current spec and escape every reserved character.
My own feeling is that this renders most things somewhat unreadable (if
that matters), e.g.,</tt>
<p><tt>sip:j.doe@big.com;maddr=239.255.255.1;ttl=15</tt>
<p><tt>becomes:</tt>
<p><tt>sip%3aj.doe%big.com%3bmaddr%3d239.255.255.1%3bttl%3d15</tt>
<p><tt>2. The other approach is to use an "envelope" syntax, like quoted-string,
i.e.,</tt>
<p><tt>hvalue = token | quoted-string</tt>
<p><tt>Now only quotes have to be escaped (as a quoted-pair), but these
seem to be relatively infrequent in most of the SIP examples I've seen.
The rule for converting an hvalue to a field-content :</tt>
<br><tt>a) if it's a token, it is the field-content</tt>
<br><tt>b) if it's a quoted-string, remove the outer quotes and convert
each contained quoted-pair to corresponding character.</tt>
<p><tt>Now if quoted-string is a problem for the reason cited earlier in
a similar discussion on parameter names and values (double quote marks
are excluded from URI's), use a single quoted string instead. Single quote
is allowed in URI's and I don't think it's used for any SIP syntax. However,
this only excludes them from the outer level syntax unless we want to exclude
quoted-strings from all of SIP overall. (Or maybe just from any SIP contained
in URL headers.)</tt>
<p><tt>In any case, approach 2. seems more attractive to me, but I'm not
sure what current practice is in this area.</tt>
<p><tt>Rick Workman</tt>
<br><tt>Nortel Networks</tt>
<br><tt>Ottawa</tt>
<br>&nbsp;
<p><tt>P.S. Minor point: In section G "Changes to Be Made", a bug is recorded
for the Call-ID header which proposes to change the field-content to token.
I agree with the intent of the proposal, but I don't think token allows
@. This may break a lot of existing implementations since the current RFC
insists on @ being present. Also note that the current spec also allows
@ in the local-id which may cause a bit of confusion, but I suspect most
implementations just treat Call-ID as a single character sequence anyway.</tt>
<br><tt></tt>&nbsp;</html>

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Please, see my question/comment bellow.

>
>
> Some additional comments:
>
> Igor Slepchin wrote:
> >
> > > 4) If the answer to the previous questions are YES, than the
> location server
> > > would return REDIRECT to addressK@domainM and PROXY to
> addressL@domainN -
> > > what is
> > > a scenario that the spec explicitly tries to avoid by enforcing that
> > > all contacts on a REGISTER must have the same 'action'.
> >
> > I guess the question is about what to do when the recursed Contact has
> > the action different from that in the first Contact. Well, that's up to
> > the proxy: it can overwrite that action or, if this turns out to be the
> > last Contact to try, it can overwrite the action from the first Contact.
>
> It should generally do whatever would happen if the domains were all
> served by proxies. So, given the description of the registrations:
>
> > userA@domainX registers with action=PROXY, contact1=userB@domainY
> > and contact2 = userC@domainZ.
> > userB@domainY registers with action=REDIRECT, contact=addressK@domainM
> > and userC@domainZ registers with action=PROXY, contact=addressL@domainN
>
>  the following would happen:
>
> 1. an INVITE arrives at proxy1 for userA@domainX. Based on its
> registrations, it proxies this to userB@domainY and userC@domainZ
>
> 2. one invite goes to domainY and one to domainZ. domainY looks up
> userB, and sends a 300 redirect with Contact addressk@domainM. domainZ
> looks up userC, and proxies to addressL@domainN.
>
> 3. the 300 redirect arrives at the first proxy, which proxies to
> addressk@domainM. The proxy for domainN receives the INVITE for
> addressL@domainN, and does whatever its configured to do.
>

I guess it is also accepted for a Proxy Server to return that 300 redirect
back to the UAC - if that is the best response received among all, instead
of recursing on it. Better still is if the proxy server can have some
internal
criteria to decide whether to recurse on the returned address or not, say,
if
the address is not a SIP URL but rather a generic URI than the server
returns
the response to the UAC, otherwise it sends an Invitation to that address.
Does that make sense?

>
> Now, should any of these domains be served by the same proxy, the
> behavior should be exactly the same as if they were not served by the
> same proxy. So, if domainX, domainY, domainZ are all served by one
> proxy, that proxy would have ended up proxying the received request to
> addressk@domainM and addressL@domainN.
>
> -Jonathan R.
>
>
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
>



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Subject: Re: [SIP] 2543 bis: Section H.1 URI Syntax/header syntax
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On Fri, May 5 2000, "Rick Workman" wrote to "SIP" saying:

> Regarding header syntax in the Open Issues section, it's suggested that
> hname and hvalue can't be *uric because of the obvious ambiguities. One
> suggestion is to just disallow  "&" and "=". Another possibility is to
> just remove reserved as an alternative in the uric rule, since the text
> states that "All URL reserved characters in the header names and values
> must be escaped", i.e., there's no point in including reserved in the
> BNF for uric, at least for this purpose.

My inclination is to just disallow "&" and "=", as well as any other
syntactially obnoxious characters such as ",".  I think it's most in the
spirit of the URI syntax.

> It seems the general intent is to allow the specification of SIP headers
> in SIP URL's. Since the message header field-name [6.6] is defined to be
> a token, this suggests that it's sufficent that hname is a token as
> well. Unfortunately, the generic definition of field-value isn't quite
> so helpful:
> 
> field-value = *(field-content | LWS)
> field-content = <the OCTETs making up the field-value and consisting
>                  of either *TEXT-UTF8 or combinations of token,
>                  separators and quoted-string>
> 
> So perhaps the first thing to do is to replace the ABNF prose rules for
> field-content and TEXT-UTF8 with something more formal, e.g.,
> 
> field-content = quoted-string | separator | token | TEXT-UTF8char
> 
> where
> 
> TEXT-UTF8 = TEXT-UTF8char | LWS
> TEXT-UTF8char = %x21-7e | %x80-ff

This isn't a complete definition of TEXT-UTF8.  Maybe this is pedantic, but
a proper BNF for the RFC 2279 definition of UTF8 would be:

TEXT-UTF8 = LWS
          | %x21-7e
          | UTF8-NONASCII

UTF8-NONASCII = %xc0-df 1UTF8-CONT
              | %xe0-ef 2UTF8-CONT
              | %xf0-f7 3UTF8-CONT
              | %xf8-fb 4UTF8-CONT
              | %xfc-fd 5UTF8-CONT

UTF8-CONT : %x80-bf


Splitting out UTF8-NONASCII from TEXT-UTF8 allows easy syntactic definitions
of such things as quoted string bodies (qdtext), which are currently
textually defined as < any TEXT-UTF8 except " or \ >:

qdtext : LWS
       | %x21 | %x23-5b | %x5d-7e
       | UTF8-NONASCII

The question still arises of how do you encode UTF8 field content in a
header parameter.  I think the only right solution is to encode first in
UTF8, then to do hex %-escaping; the alternative, %-escaping 16-bit or
32-bit unicode, is just awful.

> 1. Follow the current spec and escape every reserved character. My own
> feeling is that this renders most things somewhat unreadable (if that
> matters), e.g.,
> 
> sip:j.doe@big.com;maddr=239.255.255.1;ttl=15
> 
> becomes:
> 
> sip%3aj.doe%big.com%3bmaddr%3d239.255.255.1%3bttl%3d15

This isn't correct, because the point of escaped characters is to
distinguish *URL* syntax characters from field characters.  Thus, the URLs:

sip:j.doe@big.com;maddr=239.255.255.1;ttl=15

and

sip:j.doe@big.com;maddr=239.255.255.1%3Bttl%3D15

are *different* URLs; the second has only a single parameter, whose name is
"maddr" and whose value is "239.255.255.1;ttl=15".  I think anything we do
needs to be sure to preserve this distinction.

-- 
Jonathan Lennox
lennox@cs.columbia.edu


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Subject: Re: [SIP] 2543 bis: Section H.1 URI Syntax/header syntax
References: <3912C921.D7E42893@americasm01.nt.com> <200005051417.KAA19665@ind.cs.columbia.edu>
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> 
> This isn't a complete definition of TEXT-UTF8.  Maybe this is pedantic, but
> a proper BNF for the RFC 2279 definition of UTF8 would be:
> 
> TEXT-UTF8 = LWS
>           | %x21-7e
>           | UTF8-NONASCII
> 
> UTF8-NONASCII = %xc0-df 1UTF8-CONT
>               | %xe0-ef 2UTF8-CONT
>               | %xf0-f7 3UTF8-CONT
>               | %xf8-fb 4UTF8-CONT
>               | %xfc-fd 5UTF8-CONT
> 
> UTF8-CONT : %x80-bf

Anything formal for TEXT-UTF8-TRIM?

Maybe = TEXT-UTF8-NOLWS *TEXT-UTF8

but that doesn't express the lack of trailing LWS, although I'm not sure
this actually matters from a parsing perspective.


> The question still arises of how do you encode UTF8 field content in a
> header parameter.  I think the only right solution is to encode first in
> UTF8, then to do hex %-escaping; the alternative, %-escaping 16-bit or
> 32-bit unicode, is just awful.

http://www.ietf.org/internet-drafts/draft-masinter-url-i18n-05.txt
describes this problem and does indeed suggest that the bytes are
hex-encoded if needed. Unfortunately, it's still a pretty rough draft
and unlikely to be citable in RFC2543bis.




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On Fri, May 5 2000, "Henning Schulzrinne" wrote to "Jonathan Lennox, sip@lists.bell-labs.com" saying:

> Anything formal for TEXT-UTF8-TRIM?
> 
> Maybe = TEXT-UTF8-NOLWS *TEXT-UTF8

TEXT-UTF8-TRIM = TEXT-UTF8-NOLWS *TEX-UTF8 TEXT-UTF8-NOLWS
               | TEXT-UTF8-NOLWS
               | ; empty string

This should work.  I'm not sure if you want to allow the last one.

-- 
Jonathan Lennox
lennox@cs.columbia.edu


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> Henning Schulzrinne wrote: 
> Thus, whoever was dealing with the Mozilla folks can now point to the
> document
> http://www.isi.edu/in-notes/iana/assignments/url-schemes
> to prove legitimacy.

Done. See http://bugzilla.mozilla.org/show_bug.cgi?id=37637.
This bug has been marked "Help Wanted" by mozilla folks, so if you are
interested in implementing it, go ahead
(http://www.mozilla.org/hacking/).
I might try to do it.

Mathieu

> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> 
> 
> 
>     ---------------------------------------------------------------
> 
> Subject: RE: SIP URL scheme
> Date: Thu, 4 May 2000 18:18:37 -0400
> From: IANA <iana@ISI.EDU>
> Reply-To: iana@iana.org
> To: 'Henning Schulzrinne' <schulzrinne@cs.columbia.edu>
> 
> Dear Henning,
> 
> Thank you for notifying the IANA.  This has been added.
> 
> Best regards,
> 
> Michelle Schipper, Administrator
> IANA
> 
> -----Original Message-----
> From: hgs@cs.columbia.edu [mailto:hgs@cs.columbia.edu]On Behalf Of
> Henning Schulzrinne
> Sent: Wednesday, May 03, 2000 8:25 AM
> To: iana@ISI.EDU
> Cc: sip@lists.bell-labs.com; Mathieu Gervais (LMC)
> Subject: SIP URL scheme
> 
> Dear IANA:
> 
> ftp://ftp.isi.edu/in-notes/iana/assignments/url-schemes is missing the
> 
> sip: URL scheme defined in RFC 2543, a Proposed Standard. What is the
> process of having it added to the list?
> 
> Thank you.
> 
> Henning
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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--------------8715E4EEA33F09C9F5850A2F
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Another possibility which includes the empty string (I think this works):

TEXT-UTF8 = *(TEXT-UTF8char | LWS)

TEXT-UTF8-TRIM = *TEXT-UTF8char *(*LWS TEXT-UTF8char)

TEXT-UTF8char = %x21-7e | UTF8-NONASCII

; same as before
UTF8-NONASCII = %xc0-df 1UTF8-CONT
              | %xe0-ef 2UTF8-CONT
              | %xf0-f7 3UTF8-CONT
              | %xf8-fb 4UTF8-CONT
              | %xfc-fd 5UTF8-CONT

UTF8-CONT = %x80-bf


These are subtly different from the current TEXT-UTF8 and TEXT-UTF8-TRIM definitions in that the repeat
has been moved inside the rule, e.g., the subject header ABNF should be changed to:

Subject = ("Subject" | "s") ":" TEXT-UTF8-TRIM



Rick



Jonathan Lennox wrote:

> On Fri, May 5 2000, "Henning Schulzrinne" wrote to "Jonathan Lennox, sip@lists.bell-labs.com" saying:
>
> > Anything formal for TEXT-UTF8-TRIM?
> >
> > Maybe = TEXT-UTF8-NOLWS *TEXT-UTF8
>
> TEXT-UTF8-TRIM = TEXT-UTF8-NOLWS *TEX-UTF8 TEXT-UTF8-NOLWS
>                | TEXT-UTF8-NOLWS
>                | ; empty string
>
> This should work.  I'm not sure if you want to allow the last one.
>
> --
> Jonathan Lennox
> lennox@cs.columbia.edu
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--------------8715E4EEA33F09C9F5850A2F
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<tt>Another possibility which includes the empty string (I think this works):</tt><tt></tt>
<p><tt>TEXT-UTF8 = *(TEXT-UTF8char | LWS)</tt><tt></tt>
<p><tt>TEXT-UTF8-TRIM = *TEXT-UTF8char *(*LWS TEXT-UTF8char)</tt><tt></tt>
<p><tt>TEXT-UTF8char = %x21-7e | UTF8-NONASCII</tt><tt></tt>
<p><tt>; same as before</tt>
<br><tt>UTF8-NONASCII = %xc0-df 1UTF8-CONT</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
| %xe0-ef 2UTF8-CONT</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
| %xf0-f7 3UTF8-CONT</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
| %xf8-fb 4UTF8-CONT</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
| %xfc-fd 5UTF8-CONT</tt><tt></tt>
<p><tt>UTF8-CONT = %x80-bf</tt>
<br><tt></tt>&nbsp;<tt></tt>
<p><tt>These are subtly different from the current TEXT-UTF8 and TEXT-UTF8-TRIM
definitions in that the repeat has been moved inside the rule, e.g., the
subject header ABNF should be changed to:</tt><tt></tt>
<p><tt>Subject = ("Subject" | "s") ":" TEXT-UTF8-TRIM</tt>
<br><tt></tt>&nbsp;
<br><tt></tt>&nbsp;<tt></tt>
<p><tt>Rick</tt>
<br><tt></tt>&nbsp;
<br><tt></tt>&nbsp;<tt></tt>
<p><tt>Jonathan Lennox wrote:</tt>
<blockquote TYPE=CITE><tt>On Fri, May 5 2000, "Henning Schulzrinne" wrote
to "Jonathan Lennox, sip@lists.bell-labs.com" saying:</tt><tt></tt>
<p><tt>> Anything formal for TEXT-UTF8-TRIM?</tt>
<br><tt>></tt>
<br><tt>> Maybe = TEXT-UTF8-NOLWS *TEXT-UTF8</tt><tt></tt>
<p><tt>TEXT-UTF8-TRIM = TEXT-UTF8-NOLWS *TEX-UTF8 TEXT-UTF8-NOLWS</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
| TEXT-UTF8-NOLWS</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
| ; empty string</tt><tt></tt>
<p><tt>This should work.&nbsp; I'm not sure if you want to allow the last
one.</tt><tt></tt>
<p><tt>--</tt>
<br><tt>Jonathan Lennox</tt>
<br><tt>lennox@cs.columbia.edu</tt><tt></tt>
<p><tt>_______________________________________________</tt>
<br><tt>SIP mailing list</tt>
<br><tt>SIP@lists.bell-labs.com</tt>
<br><tt><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></tt></blockquote>
<tt></tt></html>

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I think the big picture argument you're making is that it's better to define a
reserved character set for hvalue that just removes the obvious offenders ("&",
"=", and ",") from the current definition. A couple of observations:

1. The text that says "All URL reserved characters in the header names and
values must be escaped" has me confused. Either it's just wrong, or there really
isn't much point in defining the reserved set in the definition, i.e., any
characters not in the uric set of characters must be escaped.

2. If this is true, characters not in the uric set which must escaped include
"<", ">", "\", double quote and space. These are present in SIP and could be
included in the reserved set for hvalue to improve readability, but are either
excluded or "unwise" according to the URI spec.

3. A SIP URL, including all the headers, can't contain any line breaks. This
probably shouldn't come as a surprise to anyone since most people have had to
deal with long http URLs in emails etc.


Thanks for the clarification on the TEXTUTF-8. In my opinion, we're not pedantic
enough when it comes to using ABNF. Prose rules ought to be strongly
discouraged. They are often used because or lack of understanding of ABNF or
sheer laziness. The few remaining cases are usually misguided attempts to
capture semantics in a syntax specification.

Rick


Jonathan Lennox wrote:

> On Fri, May 5 2000, "Rick Workman" wrote to "SIP" saying:
>
> > Regarding header syntax in the Open Issues section, it's suggested that
> > hname and hvalue can't be *uric because of the obvious ambiguities. One
> > suggestion is to just disallow  "&" and "=". Another possibility is to
> > just remove reserved as an alternative in the uric rule, since the text
> > states that "All URL reserved characters in the header names and values
> > must be escaped", i.e., there's no point in including reserved in the
> > BNF for uric, at least for this purpose.
>
> My inclination is to just disallow "&" and "=", as well as any other
> syntactially obnoxious characters such as ",".  I think it's most in the
> spirit of the URI syntax.
>
> > It seems the general intent is to allow the specification of SIP headers
> > in SIP URL's. Since the message header field-name [6.6] is defined to be
> > a token, this suggests that it's sufficent that hname is a token as
> > well. Unfortunately, the generic definition of field-value isn't quite
> > so helpful:
> >
> > field-value = *(field-content | LWS)
> > field-content = <the OCTETs making up the field-value and consisting
> >                  of either *TEXT-UTF8 or combinations of token,
> >                  separators and quoted-string>
> >
> > So perhaps the first thing to do is to replace the ABNF prose rules for
> > field-content and TEXT-UTF8 with something more formal, e.g.,
> >
> > field-content = quoted-string | separator | token | TEXT-UTF8char
> >
> > where
> >
> > TEXT-UTF8 = TEXT-UTF8char | LWS
> > TEXT-UTF8char = %x21-7e | %x80-ff
>
> This isn't a complete definition of TEXT-UTF8.  Maybe this is pedantic, but
> a proper BNF for the RFC 2279 definition of UTF8 would be:
>
> TEXT-UTF8 = LWS
>           | %x21-7e
>           | UTF8-NONASCII
>
> UTF8-NONASCII = %xc0-df 1UTF8-CONT
>               | %xe0-ef 2UTF8-CONT
>               | %xf0-f7 3UTF8-CONT
>               | %xf8-fb 4UTF8-CONT
>               | %xfc-fd 5UTF8-CONT
>
> UTF8-CONT : %x80-bf
>
> Splitting out UTF8-NONASCII from TEXT-UTF8 allows easy syntactic definitions
> of such things as quoted string bodies (qdtext), which are currently
> textually defined as < any TEXT-UTF8 except " or \ >:
>
> qdtext : LWS
>        | %x21 | %x23-5b | %x5d-7e
>        | UTF8-NONASCII
>
> The question still arises of how do you encode UTF8 field content in a
> header parameter.  I think the only right solution is to encode first in
> UTF8, then to do hex %-escaping; the alternative, %-escaping 16-bit or
> 32-bit unicode, is just awful.
>
> > 1. Follow the current spec and escape every reserved character. My own
> > feeling is that this renders most things somewhat unreadable (if that
> > matters), e.g.,
> >
> > sip:j.doe@big.com;maddr=239.255.255.1;ttl=15
> >
> > becomes:
> >
> > sip%3aj.doe%big.com%3bmaddr%3d239.255.255.1%3bttl%3d15
>
> This isn't correct, because the point of escaped characters is to
> distinguish *URL* syntax characters from field characters.  Thus, the URLs:
>
> sip:j.doe@big.com;maddr=239.255.255.1;ttl=15
>
> and
>
> sip:j.doe@big.com;maddr=239.255.255.1%3Bttl%3D15
>
> are *different* URLs; the second has only a single parameter, whose name is
> "maddr" and whose value is "239.255.255.1;ttl=15".  I think anything we do
> needs to be sure to preserve this distinction.
>
> --
> Jonathan Lennox
> lennox@cs.columbia.edu

--------------8860A1B6104EA9BCC86DA751
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
&nbsp;
<br><tt>I think the big picture argument you're making is that it's better
to define a reserved character set for hvalue that just removes the obvious
offenders ("&amp;", "=", and ",") from the current definition. A couple
of observations:</tt>
<p><tt>1. The text that says "All URL reserved characters in the header
names and values must be escaped" has me confused. Either it's just wrong,
or there really isn't much point in defining the reserved set in the definition,
i.e., any characters not in the uric set of characters must be escaped.</tt>
<p><tt>2. If this is true, characters not in the uric set which must escaped
include "&lt;", ">", "\", double quote and space. These are present in
SIP and could be included in the reserved set for hvalue to improve readability,
but are either excluded or "unwise" according to the URI spec.</tt>
<p><tt>3. A SIP URL, including all the headers, can't contain any line
breaks. This probably shouldn't come as a surprise to anyone since most
people have had to deal with long http URLs in emails etc.</tt>
<br>&nbsp;
<p><tt>Thanks for the clarification on the TEXTUTF-8. In my opinion, we're
not pedantic enough when it comes to using ABNF. Prose rules ought to be
strongly discouraged. They are often used because or lack of understanding
of ABNF or sheer laziness. The few remaining cases are usually misguided
attempts to capture semantics in a syntax specification.</tt>
<p><tt>Rick</tt>
<br><tt></tt>&nbsp;<tt></tt>
<p><tt>Jonathan Lennox wrote:</tt>
<blockquote TYPE=CITE><tt>On Fri, May 5 2000, "Rick Workman" wrote to "SIP"
saying:</tt>
<p><tt>> Regarding header syntax in the Open Issues section, it's suggested
that</tt>
<br><tt>> hname and hvalue can't be *uric because of the obvious ambiguities.
One</tt>
<br><tt>> suggestion is to just disallow&nbsp; "&amp;" and "=". Another
possibility is to</tt>
<br><tt>> just remove reserved as an alternative in the uric rule, since
the text</tt>
<br><tt>> states that "All URL reserved characters in the header names
and values</tt>
<br><tt>> must be escaped", i.e., there's no point in including reserved
in the</tt>
<br><tt>> BNF for uric, at least for this purpose.</tt>
<p><tt>My inclination is to just disallow "&amp;" and "=", as well as any
other</tt>
<br><tt>syntactially obnoxious characters such as ",".&nbsp; I think it's
most in the</tt>
<br><tt>spirit of the URI syntax.</tt>
<p><tt>> It seems the general intent is to allow the specification of SIP
headers</tt>
<br><tt>> in SIP URL's. Since the message header field-name [6.6] is defined
to be</tt>
<br><tt>> a token, this suggests that it's sufficent that hname is a token
as</tt>
<br><tt>> well. Unfortunately, the generic definition of field-value isn't
quite</tt>
<br><tt>> so helpful:</tt>
<br><tt>></tt>
<br><tt>> field-value = *(field-content | LWS)</tt>
<br><tt>> field-content = &lt;the OCTETs making up the field-value and
consisting</tt>
<br><tt>>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
of either *TEXT-UTF8 or combinations of token,</tt>
<br><tt>>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
separators and quoted-string></tt>
<br><tt>></tt>
<br><tt>> So perhaps the first thing to do is to replace the ABNF prose
rules for</tt>
<br><tt>> field-content and TEXT-UTF8 with something more formal, e.g.,</tt>
<br><tt>></tt>
<br><tt>> field-content = quoted-string | separator | token | TEXT-UTF8char</tt>
<br><tt>></tt>
<br><tt>> where</tt>
<br><tt>></tt>
<br><tt>> TEXT-UTF8 = TEXT-UTF8char | LWS</tt>
<br><tt>> TEXT-UTF8char = %x21-7e | %x80-ff</tt>
<p><tt>This isn't a complete definition of TEXT-UTF8.&nbsp; Maybe this
is pedantic, but</tt>
<br><tt>a proper BNF for the RFC 2279 definition of UTF8 would be:</tt>
<p><tt>TEXT-UTF8 = LWS</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | %x21-7e</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | UTF8-NONASCII</tt>
<p><tt>UTF8-NONASCII = %xc0-df 1UTF8-CONT</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
| %xe0-ef 2UTF8-CONT</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
| %xf0-f7 3UTF8-CONT</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
| %xf8-fb 4UTF8-CONT</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
| %xfc-fd 5UTF8-CONT</tt>
<p><tt>UTF8-CONT : %x80-bf</tt>
<p><tt>Splitting out UTF8-NONASCII from TEXT-UTF8 allows easy syntactic
definitions</tt>
<br><tt>of such things as quoted string bodies (qdtext), which are currently</tt>
<br><tt>textually defined as &lt; any TEXT-UTF8 except " or \ >:</tt>
<p><tt>qdtext : LWS</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | %x21 | %x23-5b | %x5d-7e</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | UTF8-NONASCII</tt>
<p><tt>The question still arises of how do you encode UTF8 field content
in a</tt>
<br><tt>header parameter.&nbsp; I think the only right solution is to encode
first in</tt>
<br><tt>UTF8, then to do hex %-escaping; the alternative, %-escaping 16-bit
or</tt>
<br><tt>32-bit unicode, is just awful.</tt>
<p><tt>> 1. Follow the current spec and escape every reserved character.
My own</tt>
<br><tt>> feeling is that this renders most things somewhat unreadable
(if that</tt>
<br><tt>> matters), e.g.,</tt>
<br><tt>></tt>
<br><tt>> sip:j.doe@big.com;maddr=239.255.255.1;ttl=15</tt>
<br><tt>></tt>
<br><tt>> becomes:</tt>
<br><tt>></tt>
<br><tt>> sip%3aj.doe%big.com%3bmaddr%3d239.255.255.1%3bttl%3d15</tt>
<p><tt>This isn't correct, because the point of escaped characters is to</tt>
<br><tt>distinguish *URL* syntax characters from field characters.&nbsp;
Thus, the URLs:</tt>
<p><tt>sip:j.doe@big.com;maddr=239.255.255.1;ttl=15</tt>
<p><tt>and</tt>
<p><tt>sip:j.doe@big.com;maddr=239.255.255.1%3Bttl%3D15</tt>
<p><tt>are *different* URLs; the second has only a single parameter, whose
name is</tt>
<br><tt>"maddr" and whose value is "239.255.255.1;ttl=15".&nbsp; I think
anything we do</tt>
<br><tt>needs to be sure to preserve this distinction.</tt>
<p><tt>--</tt>
<br><tt>Jonathan Lennox</tt>
<br><tt>lennox@cs.columbia.edu</tt></blockquote>
</html>

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Subject: Re: [SIP] 2543 bis: Section H.1 URI Syntax/header syntax
References: <3912C921.D7E42893@americasm01.nt.com> <200005051417.KAA19665@ind.cs.columbia.edu> <3912E862.2E967219@cs.columbia.edu> <200005051647.MAA21169@ind.cs.columbia.edu> <39130BEC.7273EAFE@americasm01.nt.com>
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Rick Workman wrote:
> 
> Another possibility which includes the empty string (I think this
> works):
> 
> TEXT-UTF8 = *(TEXT-UTF8char | LWS)
> 
> TEXT-UTF8-TRIM = *TEXT-UTF8char *(*LWS TEXT-UTF8char)
> 
> TEXT-UTF8char = %x21-7e | UTF8-NONASCII
> 
> ; same as before
> UTF8-NONASCII = %xc0-df 1UTF8-CONT
>               | %xe0-ef 2UTF8-CONT
>               | %xf0-f7 3UTF8-CONT
>               | %xf8-fb 4UTF8-CONT
>               | %xfc-fd 5UTF8-CONT
> 
> UTF8-CONT = %x80-bf
> 
> 
> These are subtly different from the current TEXT-UTF8 and
> TEXT-UTF8-TRIM definitions in that the repeat has been moved inside
> the rule, e.g., the subject header ABNF should be changed to:
> 
> Subject = ("Subject" | "s") ":" TEXT-UTF8-TRIM
> 
> 

I updated the spec accordingly.

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


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Henning Schulzrinne wrote:
> 
> Jonathan Rosenberg wrote:
> >
> > This is also present in rfc2543, and I believe it is an error resulting
> > from cut and paste from HTTP, which contains this paragraph verbatim.
> > There is no Content-Language header in SIP. This should be stricken from
> > rfc2543bis. Thanks for pointing it out.
> 
> Hmm. If we allow HTML or other objects in the SIP response, maybe
> Content-Language doesn't seem all that far-fetched. At least that way, I
> can automatically invoke the babelfish translator when I get the Spanish
> web-IVR menu :-)

Realistically, what are the applications for Content-Language? What
would a client do differently on receiving a Spanish version vs.
English? I see the value in Accept-Language, so that I get the right
thing back in the first place, but Content-Language?

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Francois-Xavier Guitton wrote:
> 
> Dear all,
> 
> The RFC2543 is not very clear regarding the SDP content of a 606 - Not
> Acceptable response. I think it should be useful to specify in the 606
> response all the media capabilities supported by the callee user agent.
> It will allow the caller to send directly a new INVITE with the right
> SDP description, without using supplementary OPTIONS discovery message,
> or INVITE with no SDP.

Well, the response is supposed to contain Warning headers which indicate
what the problem was. These give additional information, such as whether
one of the media streams was unacceptable, or whether there was no
common transport protocol. Right now, though, the spec doesn't seem to
say anything about placement of SDP in the response. I suppose you could
put it there, in which case it should be formatted the same way it would
for OPTIONS.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
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Phil Hoffer wrote:
> 
> Hi All,
> 
> I think we all agree that the Bis draft is infinitely more readable than the
> plain text forerunner. A big improvement.

Well, we had a postscript version for rfc2543 also, but decided not to
make it available. Unfortunately, it is very difficult to guarantee
consistency between the ASCII and PS versions. As the ASCII version is
the absolutely correct one, we stuck with that. The next SIP revision
will be the same; the final version (as will all RFCs) will be ascii
text.

> 
> As a result of the improved readability, I require a clarification to Figure 13
> in section 11, which documents a State Transition Diagram of server for
> INVITE Method.

Actually, I believe this diagram is confusing partly because it is
combining two things - call state and transaction state. The INVITE
transaction and BYE transactions are totally separate as far as
retransmissions and the like are concerned. Each has its own state
machinery. There is also a state machine for the call. This machine
tracks the INVITE transaction state, but eventually is different once
the transaction completes.


> 
> My confusion mainly centres around the Failure state and subsequent
> transitions.
> 
> 1) If the Failure state receives an ACK, a transition to Confirmed occurs.
> This doesn't make sense to me as there is now no way to revert to Initial
> state. I can see how this works for the Success state, as the BYE event,
> as a result of a hangup, causes the transition to Initial state.
> 
> Perhaps the ACK event in the Failure state should cause a transition to
> Initial state?
> 

Again, confusing because two things are being combined. The INVITE
transaction, upon moving to confirmed, should stay there for some amount
of time, around 32s. This isn't really critical. Its main point is to
make sure that any additional ACKs are treated as retransmissions. If
the server machine moved immediately from failure to confirmed and then
initial when an ACK came, the next ACK would appear as a "stray" ACK
since the transaction state no longer exists. Stray ACKs are anyway not
a big deal, but it helps to keep the state around to make sure they all
get absorbed.

> 2) A transition from Failure state to Initial state is shown on the diagram, as
> a dashed line. Unfortunately, this transition is unlabelled. I'm referring
> to the left hand dashed line, not the BYE event which joins the Success
> dashed line.

Hmm. not sure, actually. The labeling would make this appear to be
CANCEL, but thats not correct. Henning?

> 
> 3) Does the diagram need to be modified in light of changes in section
> 4.2.5 CANCEL, which talks about the issuing of a 487 (Request Cancelled)?
> Presumably the 487 response is the response to the original INVITE, and a
> separate 200 is issued as a response to the CANCEL?

Definitely. I believe it also needs to be reworked considering the
separation of transaction state from call state. The 487 response to the
original INVITE is an example of this separation.

> 
> 4) This leads me into more questions about Cancel in section 4.2.5.
> The draft states ....
> 
> "A redirect or user agent server receiving a CANCEL request responds with a
> status of 200 (OK) if the transaction exists and a status of 481 (Transaction Does
> Not Exist) if not, but takes no further action. In particular, any existing calls is
> unaffected"
> 
> What defines "transaction does not exist" from a user agent server perspective?
> Is a transaction deemed to be non-existent if the user agent server has issued
> a final response?

Transaction doesn't exist means that the UA has no current state at all
for the transaction identified by the CANCEL. If the server has sent a
final response, but knows about the transaction, its not a 481, but a
200 OK.

> 
> If so then the Failure and Success states in Figure 13, presumably should issue
> 481 (Transaction Does Exist) as a result of CANCEL request?

No.

> 
> The user agent client is then left with two scenarios:
> 
> a) Call failed at user agent server after Cancel was sent, so ACK >= 300 response
> b) Call succeeded at user agent server after Cancel was sent, so ACK the 200
> response, and issue BYE to terminate call.

Yes. Same behavior as if there was no CANCEL, pretty much. Thats the
whole idea - CANCEL "speeds things up" in getting a response from the
UAS.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Eber Mello wrote:
> 
> Please, see my question/comment bellow.
> 
> >
> >
> > Some additional comments:
> >
> > Igor Slepchin wrote:
> > >
> > > > 4) If the answer to the previous questions are YES, than the
> > location server
> > > > would return REDIRECT to addressK@domainM and PROXY to
> > addressL@domainN -
> > > > what is
> > > > a scenario that the spec explicitly tries to avoid by enforcing that
> > > > all contacts on a REGISTER must have the same 'action'.
> > >
> > > I guess the question is about what to do when the recursed Contact has
> > > the action different from that in the first Contact. Well, that's up to
> > > the proxy: it can overwrite that action or, if this turns out to be the
> > > last Contact to try, it can overwrite the action from the first Contact.
> >
> > It should generally do whatever would happen if the domains were all
> > served by proxies. So, given the description of the registrations:
> >
> > > userA@domainX registers with action=PROXY, contact1=userB@domainY
> > > and contact2 = userC@domainZ.
> > > userB@domainY registers with action=REDIRECT, contact=addressK@domainM
> > > and userC@domainZ registers with action=PROXY, contact=addressL@domainN
> >
> >  the following would happen:
> >
> > 1. an INVITE arrives at proxy1 for userA@domainX. Based on its
> > registrations, it proxies this to userB@domainY and userC@domainZ
> >
> > 2. one invite goes to domainY and one to domainZ. domainY looks up
> > userB, and sends a 300 redirect with Contact addressk@domainM. domainZ
> > looks up userC, and proxies to addressL@domainN.
> >
> > 3. the 300 redirect arrives at the first proxy, which proxies to
> > addressk@domainM. The proxy for domainN receives the INVITE for
> > addressL@domainN, and does whatever its configured to do.
> >
> 
> I guess it is also accepted for a Proxy Server to return that 300 redirect
> back to the UAC - if that is the best response received among all, instead
> of recursing on it.

Sure.

> Better still is if the proxy server can have some
> internal
> criteria to decide whether to recurse on the returned address or not, say,
> if
> the address is not a SIP URL but rather a generic URI than the server
> returns
> the response to the UAC, otherwise it sends an Invitation to that address.
> Does that make sense?

Yup. Proxying to http URLs is not that useful. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Mon May  8 02:26:33 2000
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Subject: Re: [SIP] Content-Language header
References: <200005042223.RAA22314@center.sssi.com> <39123B97.A4633380@dynamicsoft.com> <39123A5C.D9581F88@cs.columbia.edu> <3916493B.9FE39720@dynamicsoft.com>
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> Realistically, what are the applications for Content-Language? What
> would a client do differently on receiving a Spanish version vs.
> English? I see the value in Accept-Language, so that I get the right
> thing back in the first place, but Content-Language?
> 
Automatic translation is about the only application I can think of. (In
HTTP, caching is probably another reason.) However, since the client is
perfectly free to ignore this field, I don't see why inheriting this
from HTTP is bad.


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Jonathan Rosenberg wrote:

> Phil Hoffer wrote:

<snip>

> > As a result of the improved readability, I require a clarification to Figure 13
> > in section 11, which documents a State Transition Diagram of server for
> > INVITE Method.

<snip>

> > 2) A transition from Failure state to Initial state is shown on the diagram, as
> > a dashed line. Unfortunately, this transition is unlabelled. I'm referring
> > to the left hand dashed line, not the BYE event which joins the Success
> > dashed line.
>
> Hmm. not sure, actually. The labeling would make this appear to be
> CANCEL, but thats not correct. Henning?

Presumably this unlabelled event could be a CANCEL as they UAC
might have issued the CANCEL when the UAS was in Call Proceeding
state. So the >= 300 response could cross the incoming CANCEL on
the wire. Therefore a CANCEL could arrive in the Failure state. ??!!

What do you think?

Thanks In Advance

--
Phil Hoffer                         Ubiquity Software Corporation
Software Developer                  Ubiquity House
Tel: +44 (0) 1633-765600            Langstone Park
Fax: +44 (0) 1633-765601            Newport   NP18 2LH
URL: http://www.ubiquity.net




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Final minutes of SIP WG, IETF 47. Notes recorded by Tom Taylor, edited by
Dean Willis. HTML version posted to
http://www.softarmor.com/sipwg/meets/IETF47/notes/final-minutes.html. There
were two review cycles on the list.

Minutes of SIP Working Group Meetings, IETF 47. There were two sessions and
thirty drafts. The meeting was chaired by WG chairs Jonathan Rosenberg,
Joerg Ott, and Dean Willis. Tom Taylor recorded notes, and Dean Willis
edited these into minutes.

1. Agenda Bashing
=================

The proposed agenda was accepted.  Presenters from Kyoto University
asked to be added to the agenda.  The request was denied because their
draft had not been submitted by the meeting cutoff date and the agenda
was very full.

2. Status
=========

Dean Willis presented a summary of status of the WG work items,
followed by a brief discussion of potential charter items. There was
discussion as to whether the DCS drafts would be considered for
adoption by the WG. The current drafts still have RFC2026 complienace
issues preventing WG adoption as work items, but these issues may soon
be resolved. The general consensus seems that several of the DCS
drafts are likely to be adopted once the IPS is resolved.


The WG indicated consensus to adopt several tasks as chartered WG
activities, including:
SIP server features (standards track)
Call flows (informational)
Session Timer (standards track)
Reliability of provisional messages (standards track)
SIP-T efforts:
ISUP MIME types (standards)
ISUP mapping (undetermined)
Single Line Extension (undetermined)
Guidelines for Extensions.


As a general note regarding the management of the work of the WG:
Jonathan Rosenberg is attempting to assemble teams around specific work
items, with the proviso that individual participants should not be
over-committed.  The "single line extension" team is an example of this.



3. Summary of Current Discussion On The List
============================================

Jonathan Rosenberg summarized the status of discussion of a number of
current issues.

3.1 draft-nair-sip-dhcp-00.txt

No issues have been raised on the list, nor comments received from the
DHCP WG.  The fast-track procedure applied to this work appears to be
acceptable to the SIP WG.

3.2 Content In INVITE

Discussion in late December of how to include of content such as JPEG
images in INVITEs brought out a number of points:
 -- whether to include the actual content, include it indirectly through
provision of a URI, or include it indirectly as a media stream
 -- location of the content, in a header or in the body
 -- the need to identify how the content is to be used.
Jonathan perceived no consensus on these items, due to lack of comment
on specific proposals made to the list.


3.3 WWW-Authenticate Syntax

There is a proposal before the list, but no comments have been received.

3.4 tel:// vs. sip:// URLs

Jonathan emphasized the importance of the issue, discussion of which has
stalled.  He would like to see a new push to resolve it.

3.5 URL Parameters For Telephone Numbers

A couple of issues have been discussed under this heading: how to
indicate that the number shown is the result of an LNP lookup, and how
to indicate private dialing plans.  Again, Jonathan would like to see
a renewed effort to resolve these issues.

3.6 To Quote Or Not To Quote: Digest-URI

Robert Sparks commented that most implementations use quotes around the
digest-URI.  Usage for other fields is variable.  He suggests that all
fields be quoted.

3.7 Re-INVITE Glare

Jonathan noted that three proposals are on the table for handling of
coincident re-INVITEs.  Scott Petrack noted the need for a solution that
works with more than two parties.  Jonathan responded that SIP states
are shared only on a per-call-leg basis, except possibly in the case of
multicast, so multileg coincidental invites may not be an issue.

[A couple of points in Jonathan's charts were skipped because they are
being considered by the SIP security task force.]

3.8 When To Send 183 Response

Jonathan presented the rules for use of the 183 response which had come
out of discussion on the list.  A brief discussion followed, in which a
problem with the first rule was noted and resolved.

3.9 Overlap Dialling

Jonathan noted that there was no easy solution to the possibility of
forking for a call in which overlap dialling is being used.  David Oran
suggested that when the gateway rejects the first call, it should return
a contact which redirects all subsequent INVITEs to the same gateway.
Jonathan noted that this changed the meaning of the Contact: header
slightly, but its usage remains within acceptable bounds.

3.10 Early Media Criticality

Jonathan noted a continuing unresolved issue of how to distinguish
between critical and non-critical early media.

In general Jonathan intends to put out notes to the list to stimulate
further discussion of the open issues.

4. Status Of I-Ds (Jonathan Rosenberg)[charts]
======================================

4.1 Caller Preferences
 (draft-ietf-sip-callerprefs-01.txt)

Jonathan summarized a number of changes in the current draft.  There
have been no objections to the use of case-sensitive matching.  The
draft is due to go to the IESG, but Jonathan is looking for volunteeers
to check it out first.

4.2 Supported: Header
 (draft-ietf-sip-serverfeatures-02.txt)

The draft has undergone a major overhaul.  The one open point had to do
with use of the header with ACK.  This has been fixed and the draft has
been forwarded to the IESG.

4.3 Reliable Provisional Responses
 (draft-ietf-sip-100rel-00.txt)

A number of open issues still remain.
 -- cumulative acknowledgements: looking for an application to justify
inclusion of the mechanism
 -- response to PRACK: retain mechanism
 -- CANCEL for PRACK: disallow
We are looking to pass this document to the IESG as a Proposed Standard
by mid-April.

4.4 Guidelines For Authors Of SIP Extensions
 (draft-rosenberg-sip-guidelines-00.txt)


Jonathan proposed that this be a WG work item.  It would eventually
become a BCP, but not for another year.  For now it would be a living
document as the WG accumulates experience.  The sense of the meeting was
to adopt the work item, but there were mixed views on how quickly it
should be given formal status, with some arguing for near-term RFC
status.  This point will be taken to the list.  Jonathan has volunteered
to maintain the document if the WG chooses to make it a long-term
project as he proposes.

4.5 Third-Party Call Control
 (draft-rosenberg-sip-3pcc-00.txt)
This document has a number of dependencies, and various models of
control are possible.  In general the work does not involve an
extension to SIP, but only a description of the usage of its
mechanisms.  For this reason Jonathan does not propose that it be a WG
work item, but would like people to comment on it.

5. DCS Drafts
=============

5.1 Resource Reservation (Bill Marshall) [charts]
 (draft-manyfolks-sip-resource-00.txt)

This draft came about through the compromise merger of
draft-ietf-mmusic-qos-00.txt and draft-dcsgroup-sip-resource-01.txt.
The two-stage INVITE has been discarded.  SDP enhancements in support
of QOS and security negotiation include two new attributes (a=qos: and
a=security:), with syntax provided for confirmation that the requested
attribute values have been granted.  Extensions to SIP in support of
the negotiation of these attributes are covered in the same document
as the SDP changes, with AD approval. SIP extensions include use of a
header indicating that preconditions have been met.

The question was raised, whether there was any problem with using the
a=qos: attribute with provisional responses to allow the invitee to
set preconditions.  Bill made the point that the first user of the
attribute has to know whether the other end would understand it.

5.2 Caller Identity (Flemming Andreasen) [charts]
 (draft-dcsgroup-sip-privacy-01.txt)

Flemming summarized changes in the draft.  Earlier work has been
extended to recognize the possibility of boundaries between
proxies. In geneal, rather than requiring a nounary proxy to remove
calling party information, any proxy may encrypt the field and list
itself as an authenticator.  A new header, "Remote-Party-ID", has been
defined.  The set of possible operator services has been generalized.

Scott Petrack suggested that a specific tag for operator services might
not be in the spirit of SIP.  Jonathan agreed that an alternate approach
might be possible.  The basic idea, as always in SIP, is to define a
small set of reusable features rather than a variety of specialized
ones.

5.3 Distributing Call State (Bill Marshall) [charts]
 (draft-dcsgroup-sip-state-01.txt)

The mechanism has changed completely since the previous draft, in an
attempt to make it more general.  The draft now takes care to
distinguish between transaction state, connection state, and call
state.  The rules for including the "State" header have been
simplified.  There was a comment to the effect that the state
information must include the information necessary to ensure that it is
not used for a different call from the one which generated it.

5.4 Proxy-to-Proxy Extensions (Mike Mannette) [charts]
 (draft-dcsgroup-sip-proxy-proxy-01.txt)

Mike reviewed the changes from the previous draft.  LNP support has been
provided and hostport is now a required field in the "Gate" header.  The
"Billing-Info" and "Billing-Id" headers are unchanged.

5.5 Media Authorization (B. Beser) [charts]
 (draft-dcsgroup-sip-call-auth-01.txt)

This draft addresses the need for a per-call mechanism to authorize the
flow of media.  The new draft has been changed to focus only on the
client.  The former "DCS-Local-Gate" header has been renamed to
"Media-Auth".  The content has been generalized to hexadecimal.

6. Subscribe-Notify (Adam Roach)
==========================================
 (draft-roach-sip-subscribe-notify-00.txt)

Motivation: there is a need for a central definition of concepts which
have been mentioned in various documents.

The draft formalizes the methods and headers associated with
subscription and notification.  Distinctions are made between
call-related and resource-related subscriptions, with the latter
typically being of interest to third parties.  Rules are set down for
notifications and for duration of subscriptions.  The author attempted
to avoid disruption to PINT, but it might be possible to combine the
PINT work with this new material.

Scott Petrack noted that PINT had spent a great deal of time working
to make their mechanisms sufficiently generic.  He urged Adam to
review the PINT archives.  Unsubscription proved to be a tricky topic,
and PINT found it preferable to list events in bodies rather than in
headers.  Scott and Adam will discuss this further off-line.

Henry Sinnreich asked what measures had been taken to coordinate with
the Instant Messaging and Presence (IMPP) WG.  Jonathan replied that
IMPP is addressing a different problem and had already judged SIP to
be unsuitable for their purposes.  He also recalled that a general
subscription/notification BOF was held, but nothing came out of it
because the problem was too broad.

Scott Petrack noted that PINT will change if necessary to conform to
SIP.

An example subscription/notification service was presented, where the
caller is rung back when the callee becomes free.  There are open issues
where further work is required.  Jonathan proposed that another task
force be set up on the Egroup mail server, to refine the problem
statement.

Michael Thomas questioned whether SIP really needed an asynchronous
method of completing a call when the current protocol can support a
blocking approach.

7. Mobility (S. Baba) [charts]
=========================
 (draft-itsumo-sip-mobility-req-00.txt)

The draft provides requirements, open issues, and relevant work items
in SIP and other WGs.  The requirements are organized on a functional
basis.

Addressing the charts, Lawrence Conroy asked what was meant by the
requirement to "utilize" CDMA soft hand-off.  He saw this as invisible
at the IP level.  The presenter pointed out that the process might
involve multiple IP addresses for the same UA.  This would represent a
major extension to SIP.

James Polk noted that the forthcoming Location Information BOF might
be relevant to this problem.

[end of first session]

8. Review Of Charter Work Items  (Dean Willis) [charts]
===============================

Dean asked for comments on the proposed work items before the WG
chairs reviewed them and passed the results of the review on to the
list.  There were no comments.

9. INFO Method, Session Timer, 183 Session Progress (Steve Donovan)
[charts]
===================================================

9.1 INFO Method

Work on the INFO method, draft-ietf-sip-info-method-01.txt, is basically
done.

9.2 Session Timer

The Session Timer draft (draft-ietf-sip-session-timer-01.txt) underwent
significant change to reflect the new "Supported" header.  Some issues
remain before WG Last Call:
 -- refreshing of the timer.  Jonathan suggested that this be done by
re-INVITE, rather than a new method.  Agreed.
 -- Session-Expires header problematic in anything but INVITE -- will be
allowed with INVITE only.
 -- Can a proxy add Required headers? -- could end up with multiple
Required headers if the UAC also includes one.  This doesn't seem to
break anything, but there is an interaction with authorization.  In the
absence of comment from anyone but Jonathan, this should be checked out
on the list.
 -- the question was asked: what happens if the called UA includes
Session-Expires with a re-INVITE and it contradicts the one set by the
calling UA?  Steve suggested that it would be up to the implementation
to sort this out.  The list will be invited to comment.

Next steps:
 -- agreed that this becomes an official WG work item
 -- submit shortly to WG Last Call.

9.3 183 Session Progress

draft-ietf-sip-183-00.txt has a lot of dependencies, resulting in
improved understanding of the issues.  It may be used for early media
and for pre-conditions.  Summarizing the extensions involved:
 -- full session description bodies in 18x responses
 -- add the 183 response
 -- provide the "Session" header to indicate the purpose of the session
description body
The first two items are in 2543 bis already.

It is proposed that the Session header draft be refined to narrow its
focus, taking advantage of the 183 documentation.

10. Call Control (Robert Sparks) [charts]
================================

10.1 Framework For Call Control Extensions
 (draft-campbell-sip-cc-framework-00.txt)

Needs to be aligned with the guidelines.

10.2 Call Transfer
 (draft-sparks-sip-cc-transfer-00.txt)

Proposes a new TRANSFER method (not BYE ALSO), which does not alter the
original session.  Open issues:
 -- name of the method.  Jonathan interjected that he does not want to
build protocols for specific services, hence the method should have a
more abstract name and definition. Dean Willis suggested "Introduce".
 -- whether the method can carry bodies
 -- authentication.  Dean Willis suggested an "Introduced-By" header,
signed by the transferor.

10.3 Service Context -- Use of the SIP Request URI
 (draft-campbell-sip-service-control-00.txt)

This is an informational draft proposing the transmission of service
context through use of a distinctive Request-URI.  Robert provided an
example application and described the advantages of the mechanism.

David Oran was unclear on the intent of the work.  He noted that it
takes the opposite approach to that of HTML, by using the left side of
the URL rather than the right.  The syntax conflict could make the
result different depending on the server. Dean Willis pointed out that
SMTP also uses the left hand side, and that web URLs lack the @ and
therefore HAVE no left hand side.

Robert responded that the URL is intended to be an opaque entity bound
as a whole against a particular service.  David agreed that under these
conditions the format was acceptable.

The authors were encouraged to submit this work for informational
RFC status. This work was not adopted as a group effort of the
SIP WG.

10.4 Call Flow Examples (Presentd by Robert Sparks)
 (draft-ietf-sip-call-flows-00a.txt)

The draft has had a number of updates.  Please forward any further flows
to Alan Johnston (alan.johnston@wcom.com).

Jonathan noted that the document is getting too large to handle easily.
He proposes to farm out subsections to volunteers to review.  Ideally
they should simulate the flows to validate them.  Alan Johnston will
coordinate the work.

There is an issue with dependencies on work in progress.  The document
will be partitioned so that the stable part is processed first.

11. IETF SIP MIB (David Walker)
 (draft-ietf-sip-mib-00.txt)

Issue: the document is 70 pages long.  Can this be reduced?  It covers
RFC 2543 plus the INFO method and the 183 response.  Items are defined
within configuration, statistics, and notification groups for each of
the UAC, UAS, proxy, registrar, and redirect server, except that
elements common to all of these are grouped separately.  (Nothing has
been defined as yet to be specific to the redirect server.)

Concern: too many measurements.  Discussion of this brought out the
following:
 -- measurements should be defined only if there is a stated use for
them
 -- they are used as a troubleshooting tool, but a message history tool
would do a better job
David will send a note to the list to provoke further discussion.
Comments on how measurements are used will be welcome.  The note will
also raise a number of other questions for comment.

The current draft contains a couple of known errors.

12. BCP For SIP to ISUP Mapping (Gonzalo Camarillo) [charts]
 (draft-camarillo-sip-isup-bcp-00.txt)

This draft covers requirements, call flow examples, and the mapping of
parameters between the two protocols.  Gonzalo went over the changes to
the draft:
 -- new name
 -- additional extension: umbrella REQUIRE
 -- refinement of scope
 -- a discussion of SIP bridging between ISUP networks
 -- corrections to the handling of overlap signalling
 -- added examples.

He presented a walk-through of the overlap signalling case.  There is a
lot of messaging, but, unlike in the previous draft, it works.  There is
an open issue of how to control routing of overlap signalling in the
case of a stateless proxy forwarding to loadsharing MGCs.

A question was asked: how is connection redundancy handled -- for
example, UA multihoming.  What if the successive messages arrive at
different answers.  The reply was that this issue was out of scope of
the draft.  Jonathan remarked that SCTP transport for SIP is a topic
for the future.

Gonzalo concurrently presented a report on the status of the SIP-T
work.  It was noted that of the SIP-T work, the highest priority is to
obtain approval of the ISUP MIME types document. This is currently
planned for May 2000.

13. SIP Mapping Of MLPP (James Polk) [charts]
 (draft-polk-sip-mlpp-mapping-00.txt)

There is a mismatch between SIP "priority" levels and MLPP levels.
The draft proposals are to add a level and add administrative hooks.
Jonathan remarked that the standard is not the place to specify the
administrative hooks -- they belong in an RFP.

Subsequent discussion brought out two points:
 -- added priority level in RFC 2543bis
 -- additional document describing operation of MLPP.  This would cover
a number of items besides SIP, so it should not be a SIP WG document.

Jonathan also wondered whether MLPP really impacts SIP except at the
gateway, since it is concerned with preemption of limited call
resources in the face of higher-priority demand and IP resources are
not so obviously limited as circuits.  Brian Rosen disagreed, pointing
out that MLPP is not only about preeemption.

The outcome of the discussion was consensus to discuss adding another
priority level to 2543bis, or possibly adding "token" to the grammar
allowing arbitrary extension. This will require list review. James
Polk may submit a detailed fraft showing how the SIP priority levels
can be used to implement MLPP requirements.


14. Update On ISUP MIME Types (Lyndon Ong)
 (draft-ong-sip-qsig-mime-00.txt)

The draft has one open issue: the amount of ISUP version information
required. Followup is occuring in the SIP-T subteam and will be
reviewed on the main list when ready.

15. SIP and QOS (Henry Sinnreich) [charts]
 (draft-sinnreich-sip-qos-osp-01.txt)

Henry presented a list of "to do" items associated with the work.  He
hypothesized that a new WG may be required.  As evidence of this he
cited the need for a framework, within which different QOS models might
operate.

Radhika Roy mentioned that ITU-T Study Group 16 is also doing QOS work,
and the two groups could share their results.  Henry agreed that this
was possible.

Patrice Calhoun asked about the relationship between QOS and OSP (Open
Settlement Protocol, which appeared on Henry's charts).  Henry indicated
that OSP was relevant because it supports the clearing house model of
settlements, and the latter differs from the AAA WG model.

Melinda Shore remarked that overall (thinking also of DCS) there seemed
to be some strange QOS work going on, driven by charging models.  Hooks
within SIP are in scope for SIP WG discussion, but the general topic of
QOS for VoIP is out of scope.

Continuing in his presentation, Henry presented call flows showing the
tight coupling between signalling, QOS, and AAA.

Jonathan remarked again that ONLY changes required to SIP are within
the SIP WG's scope.

Dave Oran observed that DCS covers the "push model" of QOS: policy is
pushed to the enforcement point.  There is a need to develop a system
for the pull model, with open security issues standing in the
way. Dave proposed COPS-PR for this role.

Dean Willis suggested that perhaps the Transport Area WG could tackle the
larger problem.  Jonathan proposed discussing the matter with the ADs.

16. SIP Through Firewalls and NATs (Jonathan Rosenberg) [charts]
 (draft-rosenberg-sip-firewalls-00.txt)

Jonathan noted that the topic of firewall/NAT traversal is being covered
by the foglamps BOF.  The question is whether the SIP WG wants to do an
Informational RFC on the specific topic of SIP operation through these
barriers.

Dean Willis suggested that the question of whether this work belongs
in SIP is of the same order as whether MLPP does -- that is, both are
enablements for SIP within specific sectors.

There was a weak hum in favour of the work item.

17. Usage of TRIP For Gateway Registration (Jonathan Rosenberg)
 (draft-rs-trip-gw-00.txt)

This was simply an alert to the topic, which had been raised in the
meeting of the IPTEL WG the previous day.

18. Transferring User Control Information In SIP REGISTER Payloads
(Jonathan Rosenberg)
 (draft-lennox-sip-reg-payload-00.txt)

We need a way to pass scripts from the client to the server.  This
draft is the same as the original version a year ago, with minor
changes.  There are several open issues.  Jonathan asked "Should this
be a SIP work item?", which started a lively discussion.

Radhika Roy suggested there was a relationship to and potential impact
on mobility; Jonathan disagreed.  Lawrence Conroy wondered how the
scripts would be authenticated.  Dean Willis mentioned several
approaches that have been suggested.

Dean put the question: "even with the other alternatives, is it useful
to do this function via REGISTER?"  There was a slight hum in favour,
no opposition, with most registering no opinion.  The work was
accepted subject to approval on the list.

19. Finding A SIP Server With SLP (James Kempf)
 (draft-kempf-sip-findsrv-00.txt)

The intention is to add a template to the IANA service type registry.

Motivation: there are several ways to locate a SIP server through DNS --
for example, by use of a naming convention.  The method presented allows
location by characteristics.  This is useful because different proxies
in a domain can have different characteristics.  The approach taken is
to have one SLP abstract service type: "service-sip", with two concrete
service types.

Jim presented an example: find a registrar supporting CPL.

An open issue is how to handle incoming calls.  With other methods of
DNS usage, one can parse down the hierarchy of server names.  With SLP
it may be necessary to use SIP forking or multicast.

Should this be a WG work item or an individual draft?  IANA approval
requires RFC documentation and approval by Eric Guttman (servloc
Chair).  Comments on the draft can go to Jim directly or to the list.

Jonathan noted that this is one of many possible methods, but that's
OK.  There is certainly no point in discussing whether this work
should go forward as opposed to some other approach. The consensus was
to not adopt this as a WG task at this time, but to follow the work as
it matures and reconsider at a futue point.

20. CGI Interface (Jonathan Rosenberg)
 (draft-lennox-sip-cgi-03.txt)

The draft is being cleaned up.  A couple of issues have been resolved.
There are alternative methods for transfer of the CGI scripts. There
appears to be no intent to make this a WG item at this time.

21. Distributed Multipoint Conferences Using SIP (Jeff Mark) [charts]
 (draft-mark-sip-dmcs-00.txt)

Jeff provided some examples, showing manipulation of full mesh
conferences.  Is there SIP interest in further such full mesh work?

Jonathan noted that conference control is not part of the SIP WG mandate.
The problem of topology management is overly complicated because call
control should be separated from link state propagation.  The latter is not
something SIP is good at.

Jonathan will organize a first pass at rearchitecting the proposal,
but there are other approaches to multi-party conferencing that may prove
worth discussing.


--
Dean Willis
dwillis@greycouncil.com








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From sip-admin@lists.bell-labs.com  Mon May  8 12:38:43 2000
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Subject: [SIP] URI's in SIP messages
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The current SIP BNF allows the general form of URI (or URI-reference?) to
be used in the request line and various headers. Observations and
questions:

1. A valid form of URI (RFC 2396)is the empty sequence since:

URI-reference = [ absoluteURI | relativeURI ] [ "#" fragment ]

This would permit the request line and various headers (To, From,
Contact. ..) without an address specification which doesn't seem to make
much sense. RFC 2396 specifies that the empty sequence URI refers to the
start of the current document; is there any corresponding abstraction for
SIP transcations?

2. The relative URI references a resource relative to a base URI of a
hierarchial scheme. So a similar question: is this a useful concept for
SIP sessions?

3. Finally RFC 2396 states (my emphasis):
  "When a URI reference is used to perform a retrieval action on the
   identified resource, the optional fragment identifier, separated from
   the URI by a crosshatch ("#") character, consists of additional
   reference information to be interpreted by the user agent after the
   retrieval action has been successfully completed.  As such, it is not
   part of a URI, but is often used in conjunction with a URI."



So this very general URI seems to have a number of properties which don't
seem to be relevant when used in place of a SIP URL. Since generality
often has a price either in implementation or specification, i.e., lots
of semantic constraints, does it make sense to restrict the syntactic
form of URI used in place of a SIP URL, e.g., to the absolute-URI form?


Rick Workman
Nortel Networks
Ottawa

--------------68AB1D8811F55E1246432D85
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
&nbsp;
<br><tt>The current SIP BNF allows the general form of URI (or URI-reference?)
to be used in the request line and various headers. Observations and questions:</tt>
<p><tt>1. A valid form of URI (RFC 2396)is the empty sequence since:</tt>
<p><tt>URI-reference = [ absoluteURI | relativeURI ] [ "#" fragment ]</tt>
<p><tt>This would permit the request line and various headers (To, From,
Contact. ..) without an address specification which doesn't seem to make
much sense. RFC 2396 specifies that the empty sequence URI refers to the
start of the current document; is there any corresponding abstraction for
SIP transcations?</tt>
<p><tt>2. The relative URI references a resource relative to a base URI
of a hierarchial scheme. So a similar question: is this a useful concept
for SIP sessions?</tt>
<p><tt>3. Finally RFC 2396 states (my emphasis):</tt>
<br><tt>&nbsp; "When a URI reference is used to perform a retrieval action
on the</tt>
<br><tt>&nbsp;&nbsp; identified resource, the optional fragment identifier,
separated from</tt>
<br><tt>&nbsp;&nbsp; the URI by a crosshatch ("#") character, consists
of additional</tt>
<br><tt>&nbsp;&nbsp; reference information to be interpreted by the user
agent after the</tt>
<br><tt>&nbsp;&nbsp; retrieval action has been successfully completed.&nbsp;
As such, <u>it is not</u></tt>
<br><tt>&nbsp;&nbsp; <u>part of a URI</u>, but is often used in conjunction
with a URI."</tt>
<br>&nbsp;
<br>&nbsp;
<p><tt>So this very general URI seems to have a number of properties which
don't seem to be relevant when used in place of a SIP URL. Since generality
often has a price either in implementation or specification, i.e., lots
of semantic constraints, does it make sense to restrict the syntactic form
of URI used in place of a SIP URL, e.g., to the absolute-URI form?</tt>
<br>&nbsp;
<p><tt>Rick Workman</tt>
<br><tt>Nortel Networks</tt>
<br><tt>Ottawa</tt></html>

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Henning Schulzrinne wrote:
>
> In a redirect, you can play tricks such as adding new headers to the
> next, redirected call. (For example, the redirect server could provide
> the password for another call.) While I doubt that it is used right now,
> it seems like a feature worth having around.


Are these headers copied to the next request - as long as one of the
following
is not specified :
Call-ID, From, To and CSeq, Via ??

Any security issues with this ??



Thanks,
Shail




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> Realistically, what are the applications for Content-Language? What
> would a client do differently on receiving a Spanish version vs.
> English? I see the value in Accept-Language, so that I get the right
> thing back in the first place, but Content-Language?

Perhaps if you would send a SIP invitation with "Content-Language: Spanish"
to sip:support@somewhere.com, you would prefer the proxy at somewhere.com
to route your call to a support representative who spoke spanish.

-David H. Brandt




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Anoop,

Just call your proxy a firewall, and then you don't have to.
Though, be advised that the next downstream UAS will add a 'received'
parameter to the topmost 'via' so you will see all subsequent responses
to this request.

regards,
david

>
> Currently, the RFC states that every server in the path of a request must
add
> itself as a Via header.
>
> As a stateless proxy, if this restriction is removed then I can process
more
> calls with faster call setup times.
>
> So , can we make inserting itself as a Via optional for a stateless
proxy.
>
>
> Thanks,
>
> Anoop
>
>
>
>
>





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Some quick comments:

- It would be nice to align the event notification as much as possible
with PINT, since PINT is intended as a SIP profile. If their model is
broken, it needs to be fixed. (It may well may sense to extract the
event mechanism from PINT and have a common draft.)

- Expires: It might be cleaner to structure this similar to the Contact
mechanism in REGISTER. This also makes it possible to perform several
actions atomically. Something like

Event: waterboiling ;expires=3600 , dinnerdone;expires=7200

(I'm not sure that putting this is in the header is necessarily the best
solution, but that's a separate discussion.)
--
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs




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What would be the proper (or suggested) procedure to tie a SIP session to
an
RTSP session?

What I'd like to do is setup a session with SIP (via an INVITE), then
control streaming aspects via RTSP (i.e. play, pause, stop etc.).  However,
its not clear to me how I tie the RTSP session id to the previously
established SIP session.

Thanks.

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





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Dohoon Kim wrote:
>
> The caller lists the codecs avaiable in the session description in an
> INIVTE,
> then the callee lists the codecs which can be supported among them.
>
> If multiple codec is available in a media stream, such as
>
> m=audio 1 RTP/AVP 0 1
>
> which codec should be selected by caller,

Any one they want.

>
> and how can the callee know which codec is selected?

By the payload type number in the RTP packets.

-Jonathan R.

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




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Hi,

The Via grammar specified in the Columbia.edu's SIP page
(http://www.cs.columbia.edu/sip/SIPgrammar.html#via) is different from
the one in the draft. The difference is in the extension-params, i.e.
the grammar in the web specifies that extension-params apply to the
whole Via header, while the draft specifies that extension-params apply
to each Via element.

This is the one in the web page:

Via = ( "Via" | "v") ":" 1#(sent-protocol sent-by *( ";" via-params ) [
        comment ] ) *(";" extension-params)

And this is the one in the draft:

Via = ( "Via" | "v") ":" 1#(sent-protocol sent-by *( ";" via-params ) [
        comment ] )
via-params = ... | via-extension

Can anybody specify which one is correct?

cheers,
Bennylp





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Shail Bhatnagar wrote:
>
> A look at the syntax of SIP-URL shows that headers can be defined in a
SIP-URL
> itself. However, the spec also says they MUST NOT be present in From/To
and
> Request-URI. Does it mean they can be present in Contact header (the
addr-spec
> portion of Contact) ??
> I don't see any good reason for headers being allowed inside a SIP-URL
and I
> would be curious to know how many implementations actually introduce it
in
> SIP-URL ( probably the Contact header) and rely on it.

In a redirect, you can play tricks such as adding new headers to the
next, redirected call. (For example, the redirect server could provide
the password for another call.) While I doubt that it is used right now,
it seems like a feature worth having around.


>
> Thanks,
>
> --
> Best regards,
> Shail

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




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Folks,

Every once in a while, the question comes up of why SIP REGISTER is not
the right vehicle for allowing gateways to register phone prefixes with
a proxy. I believe TRIP is far better for this purpose. As it turns out,
a small subset of TRIP is needed to solve this problem, and yet its
still interoperable with full blown TRIP. Hussein Salama and I have
written an I-D that discusses this. Until it appears in the archives,
you can pick up a copy at:

http://www.cs.columbia.edu/~jdrosen/papers/draft-rs-trip-gw-00.txt

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




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To: "Igor Slepchin" <islepchin@dynamicsoft.com>
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Thanks for the pointers Igor. The documents really help.

----- Original Message -----
From: "Igor Slepchin" <islepchin@dynamicsoft.com>
To: "Sunitha Kumar" <skumar@vovida.com>
Cc: "SIPbell-labs" <sip@lists.research.bell-labs.com>
Sent: Thursday, March 16, 2000 8:26 PM
Subject: Re: BYE and record route


> It certainly should, and not only for BYE requests. This is discussed in
> http://www.cs.columbia.edu/~hgs/sip/notes.html (see the note on
> Record-Route towards the end of the document) and in
> http://www.cs.columbia.edu/~hgs/sip/drafts/draft-ietf-sip-2543bis-00.pdf
> (section 6.3.33).
>
> ---
> Igor Slepchin
>
>
> Sunitha Kumar wrote:
> >
> > It is given that the record route in 200 OK is copied into  the Route
> > headers for subsequent requests from the caller.  My question was, if
> > we needed to make the BYE from the *callee*  traverse the same set of
> > proxies, then  should this be rememberd from the INVITE request
> > obtained from the caller, or does the via field help in that.
> >
> > Thanks!
> >
> > --
> > Sunitha Kumar
> > Software Engineer
> > Vovida Networks
> > (408) 957 - 6374
> >
> >
>
>





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Folks,

Every once in a while, the question comes up of why SIP REGISTER is not
the right vehicle for allowing gateways to register phone prefixes with
a proxy. I believe TRIP is far better for this purpose. As it turns out,
a small subset of TRIP is needed to solve this problem, and yet its
still interoperable with full blown TRIP. Hussein Salama and I have
written an I-D that discusses this. Until it appears in the archives,
you can pick up a copy at:

http://www.cs.columbia.edu/~jdrosen/papers/draft-rs-trip-gw-00.txt

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

---------
This message came from the IETF IPTEL Working Group Mailing List.




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Sunitha Kumar wrote:
>
> Igor Slepchin wrote:
>
> > Sunitha Kumar wrote:
> > > In record route, it is given that : if the response contained  a
> > > contact header field, the user agent adds its content as the last
> > > route header.
> > >
> > > the question is;
> > >
> > > 1. after copying the content header field as the last record route
> >
> > > item, should the content header field be deleted?  ( it doesn't
> > make
> > > sense otherwise)
> >
> > The UAC should never copy UAS's Contact in its own future requests
> > regardless of whether Record-Route is present or not.
>
> So, then, is there some inconsistency in the RFC, that needs to be
> addressed?

No; the RFC never says to copy the Contact field from the response to
subsequent requests.

-Jonathan R.

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

Alternatively, what if we implement another header called "ResponsePath" .
 If a proxy wants to be in the path of the response it will include itself
there
 ( On the same lines of Route header which are used for being in path of
future
transactions).
The Via header could be used for loop detection.

Thanks,

Anoop




Shail Bhatnagar <shbhatna@cisco.com> on 03/17/2000 11:48:17 AM

Sent by:  Shail Bhatnagar <shbhatna@cisco.com>


To:   Anoop Tripathi/MW/US/3Com
cc:   sip@lists.research.bell-labs.com
Subject:  Re: Why must a stateless proxy insert a Via header for itself.



Anoop Tripathi wrote:
>
> Currently, the RFC states that every server in the path of a request must
add
> itself as a Via header.
>
> As a stateless proxy, if this restriction is removed then I can process
more
> calls with faster call setup times.
>
> So , can we make inserting itself as a Via optional for a stateless
proxy.
>
> Thanks,
>
> Anoop

Then how do you do loop detection ??

--
Best regards,
Shail








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Subject: [SIP] uploading services from the user's  home network to a sip server
 of the visited network within a 3G.IP environement
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Hi,

after doing a very good job by deciding to take SIP for call control for
multimedia services within 3G.IP networks, the 3G.IP operators have a lot
of open items concerning the architecture, one of them being which network
controls the services for a roaming user: the home network, the visited
network or shared control. The concrete question: how can I upload at the
begin or during the call the roamimg user's services available in the home
network into the SIP proxy of the visited network? As far as I know, the
CPL and REGISTER helps me to upload services only from the UA to the proxy.

Is anyone else working on similar issues? I know of the Telcordia mobility
efforts, but the 3G.IP requirements and architecture seem to be somehow
different. The 3G.IP operators would like to reuse their just implemented
GPRS and the GPRS mobility.

Thanks a lot
Laura




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Hi Scott,

Scott Bradner wrote:

> > The question is: is it posible to define, to any extent, a mechanism
> > capable of preventing the "hacked" terminal to bypass the proxy and
call
> > the "standard" endpoints witout being billed ?
>
> if they are not making any special use of the network (like QoS
> reservations or fancy proxy services ) why should they be billed
> in a way that is different than any other best-effort use of the
> network?
>

Some telephony applications are currently heavily billed, at a rate much
higher than the corresponding rate for pure best effort Internet access.

In some domains, network operators hold a kind of oligopoly, thanks to
government policies, that grant licencies only to a very restricted
number of
operators.
Please don' t quote me on this, but I would say that there is no real
competition between them, at least not harsh competition.

At the moment, they are deciding wheter to continue with circuit
switched
networks or whethether to adopt IP-based networks.

It is very likely that they make such a step *only* if they don' t see
their
revenues reduced (hopefully, they would like to increase them !)

A model where best effort calls are free and QOS calls are billed will
pobably
apply in the future, but not in the short run, at least IMHO.

Giuseppe






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Ken Carlberg wrote:
>
> > On your final thread, it should certainly be possible to define
> > an extension to SIP that is not useful in the PSTN; extensions
> > that are useful only within an enterprise for example should
> > be acceptable, or in the home, as well as something like MLPP
> > which is useful within a government communications system.
>
> Agreed with respect to something that may or may not be "useful".  But
the
> kicker that I'm curious about is whether the proposed specification is
> illegal (not allowed) when applied to the general public.

Here at IETF, we make protocols that run on the Internet, not the PSTN.
As there are no regulations regarding this kind of thing on the
Internet, it is not an issue for us.

-Jonathan R.


--
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
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To: James Yu <james.yu@neustar.com>
Cc: "'Vance Shipley'" <vances@motivity.ca>,
        John Mainwaring <crm312a@nortelnetworks.com>,
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We've discussed this on the list. Several folks pointed out that
European systems support overlap, and it was decided that it was not
reasonable to mandate the originating gateway to do digit collection to
convert this to en-bloc, although it could be optionally done.

-Jonathan R.

James Yu wrote:
>
> Vance,
>
> Thanks for the explanation.
>
> The key questions now are whether overlap signaling (e.g., Feature
Group-D
> over Multi-Frequency facility) is still used somewhere and if used where
the
> digit collection/enbloc signaling is performed in the existing SCN.
>
> If overlap signaling is still used (someone please confirm this and
indicate
> where it is still used) and a call is to be routed from the originating
SCN
> to the Internet, I guess that the originating GW (SCN-Internet)probably
has
> to perform the digit collection/enbloc signaling function so that overlap
> signaling does not propagate further into the Internet and/or the
> destination GW/SCN.  Otherwise, some entities in the Internet must
perform
> that function.
>
> James
>
--
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120
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Subject: Re: [SIP] URI's in SIP messages
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> The current SIP BNF allows the general form of URI (or URI-reference?) to
> be used in the request line and various headers. Observations and
> questions:
> 
> 1. A valid form of URI (RFC 2396)is the empty sequence since:
> 
> URI-reference = [ absoluteURI | relativeURI ] [ "#" fragment ]
> 
> This would permit the request line and various headers (To, From,
> Contact. ..) without an address specification which doesn't seem to make
> much sense. RFC 2396 specifies that the empty sequence URI refers to the
> start of the current document; is there any corresponding abstraction for
> SIP transcations?

No. Please god no. Obviously RFC2396 specifies a much wider and more abstract form
of URI than is commonly used in SIP. Not everything in 2396 is appropriate for 
every URI scheme. They are a set of guidelines that work more or less for the most
common URI schemes in use today: http, ftp, telnet, mailto (with some quirks), etc.

The Request-URI and Contact: URIs have different requirements for parsing than the
To/From URIs. For example, a proxy can to a degree treat the To:/From: URIs 
as opaque strings but MUST understand the syntax/semantics of the Request-URI.
This doesn't prevent you from doing rigorous parsing everywhere, but keep in 
mind that it doesn't necessarily need to be done everywhere.

> 
> 2. The relative URI references a resource relative to a base URI of a
> hierarchial scheme. So a similar question: is this a useful concept for
> SIP sessions?
>

The only contexts where I would think this might be useful would be in
1) The headers parameter of a SIP URI (for the From: and possibly To: headers)
2) A SIP "cookie" similar to an HTTP-cookie. This hasn't really been investigated
   much but would be interesting.

In other words, I don't think it is generally useful in the usual places in a SIP
message (To/From/Contact/Request-URI). But it is potentially useful at the user interface
level or as part of some service logic.

> 3. Finally RFC 2396 states (my emphasis):
>   "When a URI reference is used to perform a retrieval action on the
>    identified resource, the optional fragment identifier, separated from
>    the URI by a crosshatch ("#") character, consists of additional
>    reference information to be interpreted by the user agent after the
>    retrieval action has been successfully completed.  As such, it is not
>    part of a URI, but is often used in conjunction with a URI."
>

True, but this treated in very different ways by different web browsers. Some
treat it as part of the URI, others use it for "post-processing". 
It's not really used in a standard way for SIP, but as stated above it really is up
to the UserAgent's interpretation. THIS IS A GOOD THING.

> 
> So this very general URI seems to have a number of properties which don't
> seem to be relevant when used in place of a SIP URL. Since generality
> often has a price either in implementation or specification, i.e., lots
> of semantic constraints, does it make sense to restrict the syntactic
> form of URI used in place of a SIP URL, e.g., to the absolute-URI form?
> 

Well, if you really want to dig into this, the SIP URI as specified in 2543 is
actually a much different beast. Technically the entire SIP URL according to 2396
falls under the "opaque_part" category of absolute URIs. What does this mean? It means
that the syntax AND semantics of a SIP URI are NOT defined in 2396 at all but left
to the definition in 2543. So from this perspective, much of 2396 does not apply to sip:
URIs. I don't know if this was the intention of the SIP authors or not. From a common
sense perspective, however, a SIP URI has all the properties of the URLs as defined
in 2396: it is a transcribable uniform resouce identifier :)  

I see no benefit in restricting the syntax of a SIP URI. If you "ignore" the fact that
the SIP URI should be treated as opaque, you can parse the SIP URI just like an 
http, ftp, telnet, etc. URI. In this sense, it would actually place more of a burden
on an implementation to prescribe special syntax restrictions on a SIP URI. Furthermore,
I'd prefer to leave open the possibility for future SIP extensions which might need 
something like a relative SIP URI.

> 
> Rick Workman
> Nortel Networks
> Ottawa

--
Sean Olson <sean.olson@ericsson.com>
Ericsson Inc.


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Anders Kristensen wrote:
>
> "K. K. Ramakrishnan" wrote:
> >
> > I also appreciate the need to not have special casing for methods -
> > but is this extra messaging coming about because we chose to
> > have the PRACK as a "method"? On the issue of backwards compatibility,
> > couldn't we say that proxies shouldn't be creating transaction state
> > for things it doesn't understand?
>
> This proposal is itself not backwards compatible.  Also, it doesn't seem
> quite right - it would mean, for example, that proxies cannot fork on
> requests they do not know. This seems like a big limitation.

I agree completely here with Anders. The spec clearly defines handling
for unknown requests as if they were BYE.

As one of the design principles of SIP, we've really tried to make it a
fairly general purpose protocol. Many of the features and capabilities
have broad applicability and usage. Often, the result of this kind of a
decision is a tradeoff- messages are a little bigger, or more
information is sent, when its not always needed. I'd really like to keep
with that design philosophy. Having special case methods to cover every
new case, and requiring tons of Proxy-Requires to make it all work, ends
up resulting in a tangled mess of a protocol that doesn't interoperate.
Please, lets stick with the request-response model for all new methods.

-Jonathan R.

--
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
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To: "'Kundan Singh'" <kns10@cs.columbia.edu>
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Subject: [SIP] RE: SIP & RTSP interworking?
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I like this a lot better, but the problem I have is this, we'd like for
BOTH
sessions to be active at the same time and both of your proposals require a
teardown of the SIP session.

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

> -----Original Message-----
> From:   Kundan Singh [SMTP:kns10@cs.columbia.edu]
> Sent:   Wednesday, March 15, 2000 3:10 PM
> To:     Linden deCarmo
> Cc:     'Dean Willis'; sip@lists.research.bell-labs.com
> Subject:     RE: SIP & RTSP interworking?
>
> --
> Kundan Singh     http://www.cs.columbia.edu/~kns10
>
> > This is an interesting suggestion, but the scenario I'm operating under
> is
> > this:
> >
> > UAC<->UAC (in this case the UAC really is an RTSP server), both must
> > understand SIP and RTSP.  I need a way to tie the sessions between two
> > distinct protocols.  Is a viable alternative to embed the RTSP SETUP
> message
> > as a SIP body for the INVITE?  This would clearly associate the
> sessions,
> > but I hate to have a proprietary solution that wouldn't work with other
> > products that understood SIP & RTSP.
>
> In that case, the UA may return the RTSP Contact: URI in (say) 3xx
> response.
>
>  UA1          UA2
>      INVITE
>  -------------->
>
>    3xx Moved
>    Contact:rtsp://server/resource
>  <-------------
>
>    SETUP rtsp://server/resource
>  -------------->
>  . . .
>
> Since UA here understand both SIP and RTSP it should not be
> a problem.
>
> If the SIP call is already established, and UA2 wants to
> switch to RTSP,  then (IMO) it may use blind call transfer
> (BYE,Also:rtsp://...) to ask UA1 to connect to RTSP
> server of UA2. Comments ??
>




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"Hussein F. Salama" wrote:
>
> > In any case, determination of where the NPDB query is done is something
> > I don't think is within the scope of iptel to determine. All we need to
> > figure out is what kind of numbers are distributed in TRIP. Near as I
> > can tell, since LRNs still refer to the prefix of numbers served by a
> > switch, doing what we were doing all along works just fine.
> >
>
> Yes what we're doing is OK, because the LRN is part of an office's
> NPA-NXX prefix. But we may have to add a new field to the
> ReachableRoutes attribute to mark a prefix as portable. A prefix is
> marked as portable if at lease one number has been ported out of this
> prefix.

Hmm, why is that? Are you basically saying that if a prefix is marked as
portable, an LS must do an LNP query before routing to numbers in that
prefix? Then, if marked as not portable, there is no need for queries?
If so, its an interesting idea. It solves the problem of N-1, which,
from what I understand, was done in part because only carriers near the
destination would need to install triggers to perform queries on
numbers. However, if information on what prefixes are ported is
distributed in TRIP, this information is effectively dynamically
distributed. This means its equally easy for the query to be done at the
first LS as the last.

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

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Hello!

The registration for the 3rd SIP bakeoff is closed!

The popularity of this event has by far exceeded expectations.
Instead of having 15-17 teams as we expected, there are 33 teams
now on the final list and nearly 100 people.

Here is the final list of participating teams:

Name                Team Size  Contact
---------------------------------------
Nortel-1             5          orton@nortelnetworks.com
Pingtel              2          dpetrie@pingtel.com
Nortel-2             2          cjessen@nortelnetworks.com
Mediatrix            2          etremblay@mediatrix.com
Netspeak             2          noreilly@netspeak.com
Telogy               1          wkwok@telogy.com
Nuera                4          cteoh@nuera.com
Catapult             4          terry@catapult.com
Ericsson-2           3          hans@erix.ericsson.se
Mitel                2          Ashok_Ganesan@Mitel.COM
Agilent              2          douglas_carson@agilent.com
Dynamicsoft          3          vpatel@dynamicsoft.com
Broadsoft            4          joyce@broadsoft.com
VTEL                 2          rkrishna@vtel.com
Delta                3          nurban@delta-info.com
Radcom               2          mwinslow@radcomusa.com
E*Club               2          jon2@andrew.cmu.edu
8x8                  4          artru@8x8.com
Helsinki Tech U.     2          jose@tct.hut.fi
Columbia U.          3          hgs@cs.columbia.edu
HP Labs              1          ak@hplb.hpl.hp.com
Indigo               3          eb@indigo-software.com
OZ.com               2          jii@oz.com
Vovida               3          ldang@vovida.com
Cisco                3          manojb@cisco.com
IPCell               2          alex@ipcell.com
FacetCorp            5          clark@facetcorp.com
MCIW-3               4          kelvin.porter@wcom.com
3Com-1               2          Jerry_Mahler@mw.3com.com
MCIW-1               3          mohammad.vakil@wcom.com
Ericsson-1           3          adam.roach@ericsson.com
3Com-2               2          Ravandhu_hariram@3com.com
MCIW-2               5          steven.r.donovan@wcom.com
-----------------------------------------
TOTAL:  92 participants in 33 teams


Have a nice thanksgiving everyone!!

Regards,
Ulf


--
Ulf Andersson
Ericsson Inc          +1 972 583 7537        ulf.andersson@ericsson.com



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


     Title          : Event Notification in SIP
     Author(s) : A. Roach
     Filename  : draft-roach-sip-subscribe-notify-00.txt
     Pages          : 8
     Date      : 07-Mar-00

This document describes an extension to the Session Initiation
Protocol (SIP) [1] . The purpose of this extension is to provide
a generic and extensible framework by which SIP nodes can request
notification from remote nodes indicating that certain events
have occured.
Concrete uses of the mechanism described in this document may be
standardized in the future.

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

Internet-Drafts are also available by anonymous FTP. Login with the
username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
     "get draft-roach-sip-subscribe-notify-00.txt".

A list of Internet-Drafts directories can be found in
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Internet-Drafts can also be obtained by e-mail.

Send a message to:
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NOTE:     The mail server at ietf.org can return the document in
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     a MIME-compliant mail reader.  Different MIME-compliant mail readers
     exhibit different behavior, especially when dealing with
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Below is the data which will enable a MIME compliant mail reader
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Jonathan Rosenberg wrote:
>

>
> It would be really nice, however, to know ahead of time whether all
> parties supported transfer (and any other mechanisms to be defined). The
> Supported extension draft defines a way in which the UAC would indicate
> to the UAS that it knows the new mechanism, but that draft does not
> mandate that a UAS return a Supported header in the response to any
> method (only OPTIONS). We can add this, and say that if a UAS receives a
> request (with any method) with a Supported header, it should place a
> Supported header in the response and list those features it knows. This
> would allow a caller to know whether the called party supports transfer
> before trying it. Thoughts?

It might make sense to have both Allow and Supported in responses other
than OPTIONS. In this case, Allow seems the more appropriate solution,
given that it's a new method and that having two mechanisms to ask for
the same thing seems a bad design. Thus, suggesting that final responses
include Allow doesn't break anything (a UAC could have a simple logic
that did "if response didn't have Allow, send an OPTIONS just in case"),
but it does cut down on traffic. It's much nicer if the UAC can gray out
a Transfer button rather than pop up a failure note after the caller has
said "let me transfer you...".


> The document says TRANSFER can't have a body. Why not? I can think of
> some cool uses for this down the road. Why rule it out?

If nothing else, one can always use it for "if you get dropped, here's
the URL to click on to call me back...".

>
> The draft says:
> > UA receiving a well-formed TRANSFER request SHOULD request approval
> >    from the user to proceed. In the absence of that request, or upon
> >    receiving approval from the user, the UA MUST submit an INVITE to
> >    the resource identified by the Transfer-To: header using the Call-ID
> >    from the TRANSFER request.
>
> This thing about the Call-ID being copied from the TRANSFER seemed odd.
> Why is that? If you wanted to accomplish this, it seems better to have
> the URL in the Transfer-To header contain the Call-ID as a URL
> parameter. In other words, why not do:
>
> Transfer-To: sip:user@host?Call-ID=9nasd09asd--asdasd98ays

Somehow, the recipient of the transfer request must know which call id
this belongs to. It can either be a new call (when TRANSFER is really
acting as a 'create call leg, please' request) or an existing call
(standard transfer). Why create a new call-id if it relates to an
existing call?

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




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I know that a number of companies are interested in hiring summer
interns for SIP-related topics. I also know there are some students
looking for such opportunities. Since this mailing list is not really
appropriate for such topics, I've started an informal list at
http://www.cs.columbia.edu/sip. Comments, additions and suggestions (to
me, not the list) are welcome.
--
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs




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Rohan Mahy wrote:
>
> Hi,
>
> At 12:09 PM 3/9/00 , Jonathan Rosenberg wrote:
> >All requests are responded to. period.
>
> I never debated this.  The user agents will always have an opportunity to
> respond.
>
> I agree with kkrama@research.att.com, that proxies shouldn't keep state
for
> methods they don't understand.  They can simply forward messages for
these
> unknown methods on to the user agents or the next proxy.

If you want to build your proxy that way, fine. But I think it is asking
to much to mandate that unknown methods be forwarded statelessly. The
spec as written does not mandate this.


> So what I hear Jonathan saying is that because some proxies MIGHT keep
> state for new methods, then we need 200 OK, so that we don't break them.
> Correct?  I am personally curious how many proxies would break.
>
> Also, I hear Jonathan saying that new methods should be processed in the
> same way, and that this encourages simplicity.  Aside from the comment
> above, is there another motivation for this?

Yes. It leads to a clean architectural implementation. It means I can
build a base transaction mechanism that handles retransmissions and
provisional responses and stuff without detailed message semantics, and
then build semantic processing on top of this. It is this same
motivation which has led me to conclude that transactions must be kept
independent; so that, for example, if I send a BYE for an INVITE that
hasn't been answered, the UAS should still answer the INVITE.

The issue of allowing stateful proxies for unknown methods is a big one
as well. It means that SIPs routing and searching capabilities can be
reused for other services (i.e., other methods) with existing proxies. I
think this is not to be dismissed so lightly.

While there are clear advantages to keeping the 200 OK to PRACK, the
only advantage I have heard to date for dropping it is message
efficiency. In general, I firmly believe that generality and broad
applicability are far more important design goals than message
efficiency. I discuss this in more detail in the draft I recently
submitted entitled "Guidelines for Authors of SIP Extensions".

-Jonathan R.

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





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Folks,

I've just posted an individual Internet Draft which may be of interest.
Its entitled "Guidelines for Authors of SIP Extensions". Its discusses
issues that writers of SIP extensions should consider while developing
them and writing them up. It also gives some guidance on what things
make good SIP extensions, and what things do not. Until it appears in
the archives, you can pick up a copy at:

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-sip-guidelines-00.txt


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




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For those who may have missed it, here is the official announcement
of the BOF.

Dave Walker
SS8 Networks
Ottawa, Canada

-----Original Message-----
From: mbeaulie@cnri.reston.va.us [mailto:mbeaulie@cnri.reston.va.us]On
Behalf Of ietf-secretariat@ietf.org
Subject: 47th IETF: SIP323 BOF


SIP and H.323 Interworking BOF (sip323)

Monday, March 27 at 1300-1500
=============================

CHAIRS: Joon Maeng <joon_maeng@vtel.com>
        Dave Walker <drwalker@ss8networks.com>

DESCRIPTION:

There are two competing standards in the industry for establishing and
managing real-time interactive media communications over IP.  One is the
proposed IETF standard, Session Initiation Protocol (RFC 2543), and the
other is the ITU recommendation, H.323.  Although they use the same
protocol (RTP) for media streaming, they cannot establish calls between
them.  In the absence of any formal interoperability standards, many
vendors are beginning to develop proprietary application gateways for
interworking between SIP and H.323.

In order to provide interoperability between the two standards, it is
urgent to define a common interworking specifications for the industry.
The recognition of the need for this interworking specification has lead
to calls for its development from both manufacturers and service
providers, individually and through industry consortia (such as IMTC)
and other bodies concerned with interoperability (such as ETSI).

In this BOF, we are going to discuss SIP and H.323 interworking issues,
execution plans and the scope of the work.

AGENDA:

- Do we need SIP-H.323 interworking?
- Coordination with other groups?
- Scope of interworking
- Outcome of the work?
- Suggestions?

Helpful reading:
    - draft-singh-sip-h323-00.txt
    - SIP: Session Initiation Protocol (RFC 2543)
    - SDP: Session Description Protocol (RFC 2327)
    - ITU recommendations: H.323, H.225.0, H.245






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Anders Kristensen wrote:
>
> > 2. Lets keep transactions independent. We went through a similar thing
> > with CANCEL, and the state machines were confusing until we decided to
> > make CANCEL a mini-transaction that does nothing but ask the UAS to
> > quickly answer the request. I think we will see confusion in the PRACK
> > processing if its operation becomes dependent on events in other
> > transactions.
>
> It is certainly simpler to just always complete according to the
> original plan, so to speak. Interestingly, though, one might have
> thought it to be OK for the UAS to stop retransmitting reliable,
> un-PRACK'ed responses when it sends a final response. The theory here
> would be that this is so because either
>   1) the reliable response has been lost before reaching the UAC, and so
> there's no observable difference anywhere in the network from it never
> having been sent, or
>   2) the UAS did receive the reliable response so no retransmission are
> needed anyway.
>
> The only snag I can see is that some proxy somewhere may in fact be
> clever and act upon certain reliable responses, thus destroying
> assumption 1) above, so it would appear to be prudent for the UAS to
> retransmit as per the draft even in this case.

Its not so much the proxy, as the UAC. If the UAS sent the response
reliably, its probably because it contained some information that was
important for the UAC to see. Otherwise, it wouldn't have sent it
reliably in the first place.

-Jonathan R.

--
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Subject: [SIP] Re: Why do we need another header?   Re: How to put a Kerberos
 ticket in  SIP messages?
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Before jumping into the syntax issues, I'd like to understand exactly
whats going on here. Integration of Kerberos with SIP is a fairly big
topic. There has been some discussion of this for establishing security
associations between clients and local proxies as an alternative to IKE.
You are proposing a different use yet, where it seems a redirect server
is acting as a KDC, handing out tickets in SIP redirects. Lots of issues
here, syntax is the least interesting one, IMHO. How is the client
authenticating itself to the redirect/KDC in the first place? Does a
client cache tickets and try to use them later when connecting to a
server it was redirected to previously? What if the server thats
redirected to, with a ticket, is yet another redirect server?

I'm not saying I don't like this idea (in fact, I think its a very
interesting one indeed), but I'd rather discuss higher level issues
first. Actually, I'd love to see an I-D first to give a context for the
discussion.

-Jonathan R.

Bryan Byerly wrote:
>
> So I expect a few people to come back and ask why not reuse an existing
> header.
>
> I'm trying to achieve a flow where a SIP proxy hands back a Kerberos
> ticket to a client.  The client, in turn can use this ticket for
> authorization to receive a service.  (Yes, alternatively the client could
> go get his own ticket (if he knew the IP address of the KDC and was an
> authorized Kerberos client), and this follows along with moving
complexity
> out to the endpoints.  However, I'm trying to initially simplify things
> for dumber, less complex SIP clients.  (and yes, I agree that SIP clients
> could eventually go get their own tickets)).
>
> Let's go through the list of existing headers and why we can't use them:
>
> 401 WWW-Authenticate: header and Authorization: header
> Using 401's WWW-Authenticate: header and the Authorization: header are
> used for end-to-end authorization.  So, using WWW-Authenticate is not
> appropriate for passing back a Kerberos ticket.
>
> 407's Proxy-Authenticate: header and Proxy-Authorization: header
> Using 407's Proxy-Authenticate: header and the Proxy-Authorization:
header
> are used to authorize a SIP client to transit through a SIP proxy.
> Proxy-Authorization: is specified by a SIP client to provide credentials
> such that he will be authorized by the SIP proxy.  So, I don't think its
> appropriate to overload Proxy-Authorization: to mean "here's an
> authorization that the SIP proxy is passing to the SIP client".
>
> Response-Key:
> This header is used by a client to specify a key which it requests the
> called user agent should use to encrypt a response.  So, its not
> appropriate to use this header to pass a ticket.
>
> Thanks again for your help and suggestions!
>
> Bryan
>
> Bryan J. Byerly
> byerly@cisco.com
>
> Bryan Byerly wrote:
>
> > Hi guys,
> >
> > I'm trying to understand how best to place a Kerberos (v5) ticket in
> > SIP messages (i.e. either in SIP request or SIP response).
> > (If someone already has such as draft, please forward me a copy.)
> >
> > I can think of two possible approaches:
> > 1) Put it in a SIP header:
> > 2) Put it in the SDP
> > other ideas?
> >
> > 1) Thoughts on putting it in a header
> > So, a Kerberos ticket is ASN.1 DER encoded.  Is there any reason we
> > couldn't just stick the binary representation of the ASN.1 DER encoded
> > object in a header?
> > Something like:
> > ticket=fj7d543ghklc8654f7...
> > (Yes, you're right, that's not a real ticket).
> >
> > So, I'm leaning towards a more generic header name like:
> > Service-Authorization=[service name], [authorization-type],
> > encoding=[encoding-type], ticket={octet string}
> > authorization-type=kerberos5
> > encoding=octet-string
> >
> > b) Thoughts on putting it in the SDP
> >
> > Since a Kerberos ticket is ASN.1 DER encoded, it occurs to me that we'd
> > need a email-safe version of the Kerberos ticket if we were to place it
> > in the SDP.
> >
> > Can anyone point me to an existing draft/rfc which converts a Kerberos
> > key to an email-safe version.  I guess a draft on converting an
> > arbitrary octet-string to an email-safe version would be fine too.
> >
> > P.S.
> > For fun, who has a suggestion for which SDP abbreviation should be used
> > for a Kerberos ticket?  (It appears all the obvious ones are taken).
> > taken: a, b, c, e, i, k, m, o, p, r, s, t, u, v, z
> > remaining: f, g, h, j, l, n, q, w, x, y     (Please correct me if some
> > of these is already taken).
> >
> > Thanks for your help!
> >
> > Bryan
> >
> > Bryan J. Byerly
> > byerly@cisco.com

--
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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What you are describing is actually something different still.
Accept-Language describes the language that may  be used for the content
in a response. This has nothing to do with the language an operator you
might be connected to is able to speak; for that, the caller preferences
extension is targeted (in fact, it has preferences for languages
spoken). This is also different from Content-Language, which describes
the language of the html or text in a SIP response. As Henning has
pointed out, the only real use for this is some kind of automated
translation systems.

I know its optional and not hard, but I would rather not add more stuff
to an ever-more-complex protocol unless there is a very well defined
need. Since there is realistically no useful application right now of
Content-Language, I'd rather not add it.

-Jonathan R.

ozsip@oz.com wrote:
> 
> > Realistically, what are the applications for Content-Language? What
> > would a client do differently on receiving a Spanish version vs.
> > English? I see the value in Accept-Language, so that I get the right
> > thing back in the first place, but Content-Language?
> 
> Perhaps if you would send a SIP invitation with "Content-Language: Spanish"
> to sip:support@somewhere.com, you would prefer the proxy at somewhere.com
> to route your call to a support representative who spoke spanish.
> 
> -David H. Brandt
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
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--------------C7B188F4B318E3013FA2FCE1
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Hi Jonathan

Can we at least draw some consensus on the 'Meesage Waiting Indication(MWI)'
event in Voice mail context ?

Further, in one of the threads on this list, Henning says that we must adopt
drafts/discussion on PINT to decide on MWI.

Also, there is some concern that whether SUBSCRIBE -  a new SIP method
should be used for sending subscription requests or is REGISTER a good way of
doing that ?

My concern is that very soon you may have interop problems with SUBSCRIBE/
REGISTER method being used by vendors in different ways .

Is SUBSCRIBE/NOTIFY the de facto way for MWI implementation ?

Is a voice mail server supposed to act as  a UA keeping track of dynamic
subscription requests all the time and updating its database on time
or should it depend on the proxy /registrar to take the burden of MWI
by using the Contact/Registration details of the UA ?

Comments welcome...

thanks
-manoj



Jonathan Rosenberg wrote:

> I think we are once again jumping the gun a bit. There are some much
> bigger issues to address with a generic sip subscription/notification
> mechanism:
>
> 1. what entities are generating the notifications? It can be a UA, or it
> could be a proxy or registrar.
>
> 2. What are the types of events we want to support subscriptions for? I
> know Adam has tried to keep that the subject of future extensions.
> Problem is, this issue has a strong impact on the addressing, event
> description and scaling capabilities. I don't think we can leave this to
> future extensions without having the "base" version have nothing but two
> methods. That doesn't really help much. I think we can define reasonably
> broad classes of events. Certainly we aren't interested in supporting
> events like "let me know when the coffee machine is done".
>
> 3. To whom does one address a subscription? If we address it to a user,
> what does that mean? Are we asking for a subscription to that users
> registration state?
>
> Also, before we get to far with architecture and design issues, there is
> the question about whether this concept, in general, is something the
> working group is interested in pursuing.
>
> -Jonathan R.
>
> archow@hss.hns.com wrote:
> >
> > Hi,
> > responses embedded and preceded by ARC>
> >
> > >The draft does not mention what happens when multiple SUBSCRIBE requests
> > >arrive from the same user. In REGISTER,
> > >     they replace old ones if the SIP url is same and additive otherwise.
> > >However, in SUBSCRIBE, I think it would make sense to make it additive
> > >only, since the same SIP url can subscribe to multiple events using  diff.
> > >subscription messages. Is this correct ?
> >
> > That sounds good, except possibly for the same "call leg". I think the
> > part that needs to be clarified is: if a user sends a SUBSCRIBE
> > which has the same Call-ID, From, and To as a previous request,
> > does it add events or override them?  I'm leaning towards overrriding
> > events, since there is no easy way to remove events otherwise. Note well
> > that this wouldn't apply to other subscriptions with different Call-IDs,
> > which would (of course) remain additive.
> >
> > ARC> Somehow, I feel Call-ID and suscription should not be tied together.
> > This is primarily
> > ARC> because callid is generated by a terminal and a user has no control
> > over it.
> > ARC> However, a subscription is for a user, not a terminal. so if I
> > subscribe to a set of services
> > ARC> today, if I travel tomorrow, I want to walk into a Cyber Cafe, log in
> > using some applet based
> > ARC> sip client that will generate a different Call-ID from my old desktop
> > one an still be able to
> > ARC> access my old subscription list and add/del from it without having to
> > know the Call-Id - which
> > ARC> in many cases, I never get to see as a user.
> > ARC> but the moment the backend notifier works on Call-ID - then I would
> > never be able to 'Add' to
> > ARC> my list since Ill be using a different Call-ID in all probability,
> > neither will I be able to
> > ARC> somehow retrieve the old one, since match will never be true.
> > ARC> This is why I feel subscription should not be linked to a call - id.
> >
> > (Incidentally, I like your example. I have recently had
> > discussions with several different groups of people who
> > wanted to know how one would go about, for example, having
> > a light blink on your phone and/or have a distinctive dial
> > tone when a message is waiting).
> >
> > ARC> Well, we actually had exactly the above requirement once, and we did
> > it my adding
> > ARC> an Event like field with more or less the same semantics as we
> > discussed above.
> >
> > >Event =  1#(eventid ; duration)
> > >event = token       // Does it need to be broken ?
> > >duration = deltaseconds | SIP-Date
> >
> > I don't really like this. Beyond the problems with using
> > commas for separators *and* having commas appear in
> > the duration (if the SIP-Date format is used), I'm trying to
> > keep it as congruent as possible with REGISTER, since the
> > semantics for that are already well defined -- and code is
> > already written to handle it.  For the above situation, I'd
> > propose that you either: a) send all three REGISTER messages,
> > with different Call-IDs, in a single UDP packet, or b) send
> > a single REGISTER for all three events with an expires of 2
> > hours. At the end of the 2 hours, re-SUBSCRIBE for just B
> > and C (expires in one hour), and one hour later, re-SUBSCRIBE
> > for just C. (A variation on this theme would be to subscribe
> > for 4 hours, override it in 2, and then again one hour later;
> > I prefer this method, since it eliminates the chances of missing
> > events -- although it counts on the addition vs. replacement
> > semantics I describe above).
> >
> > ARC> I assume where ever you mentioned REGISTER above , you meant SUBSCRIBE
> > ARC> Again, solution 1 is to me, a kind of loose workaround. And the other
> > solutions
> > ARC> force the client to maintain time differentiators actively since it
> > must physically
> > ARC> maniuplate subscriptions to implement the common requirement above.
> > Rather, The
> > ARC> Notifier is actually already implementing a timer list which can
> > notify users
> > ARC> based on differential time values maybe, so why force the client into
> > maintain the
> > ARC> differential list also ? To me it seems to be much cleaner for the
> > client to specify
> > ARC> it once, and forget about it, as the notifier server is there just to
> > remind him anyway
> > ARC> so why bog him down with keeping track of such requests ?
> > ARC> The only -ve point seems to be parsing for commas (which is a non
> > issue for good parsers)
> > ARC> - but if that really is a problem, maybe we could work at a simpler
> > expiry type ?
> > ARC> However, I dont feel parsing should be the reason to adopt the
> > alternate methods suggested
> > ARC> above. What do you/others feel abt it ?
> >
> > I must have been somewhat unclear. I am not proposing that one
> > use REGISTER itself to cancel a subscription; merely that one
> > use the same *mechanism*, with a SUBSCRIBE method. For
> > example, let's say that I want to cancel the subscription(s)
> > above. I could do so by sending:
> >
> > ARC> Sorry, I misinterpreted, then. Seems fine to me.
>
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com

--
Manoj Bhatia                           |        |         |
manojb@cisco.com                       |       :|:       :|:
7025 Kit Creek Road, P.O. Box 14987    |     :|||||:   :|||||:
Research Triangle Park, NC 27709       |   .:|||||||:.:|||||||:.
919-392-3873      fax: 919-392-6801    |  C i s c o S y s t e m s



--------------C7B188F4B318E3013FA2FCE1
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Hi Jonathan
<p>Can we at least draw some consensus on the 'Meesage Waiting Indication(MWI)'
<br>event in Voice mail context ?
<p>Further, in one of the threads on this list, Henning says that we must
adopt
<br>drafts/discussion on PINT to decide on MWI.
<p>Also, there is some concern that whether SUBSCRIBE -&nbsp; a new SIP&nbsp;method
<br>should be used for sending subscription requests or is REGISTER a good
way of doing that ?
<p>My concern is that very soon you may have interop problems with SUBSCRIBE/
<br>REGISTER method being used by vendors in different ways .
<p>Is SUBSCRIBE/NOTIFY the de facto way for MWI implementation ?
<p>Is a voice mail server supposed to act as&nbsp; a UA keeping track of
dynamic
<br>subscription requests all the time and updating its database on time
<br>or should it depend on the proxy /registrar to take the burden of MWI
<br>by using the Contact/Registration details of the UA ?
<p>Comments welcome...
<p>thanks
<br>-manoj
<br>&nbsp;
<br>&nbsp;
<p>Jonathan Rosenberg wrote:
<blockquote TYPE=CITE>I think we are once again jumping the gun a bit.
There are some much
<br>bigger issues to address with a generic sip subscription/notification
<br>mechanism:
<p>1. what entities are generating the notifications? It can be a UA, or
it
<br>could be a proxy or registrar.
<p>2. What are the types of events we want to support subscriptions for?
I
<br>know Adam has tried to keep that the subject of future extensions.
<br>Problem is, this issue has a strong impact on the addressing, event
<br>description and scaling capabilities. I don't think we can leave this
to
<br>future extensions without having the "base" version have nothing but
two
<br>methods. That doesn't really help much. I think we can define reasonably
<br>broad classes of events. Certainly we aren't interested in supporting
<br>events like "let me know when the coffee machine is done".
<p>3. To whom does one address a subscription? If we address it to a user,
<br>what does that mean? Are we asking for a subscription to that users
<br>registration state?
<p>Also, before we get to far with architecture and design issues, there
is
<br>the question about whether this concept, in general, is something the
<br>working group is interested in pursuing.
<p>-Jonathan R.
<p>archow@hss.hns.com wrote:
<br>>
<br>> Hi,
<br>> responses embedded and preceded by ARC>
<br>>
<br>> >The draft does not mention what happens when multiple SUBSCRIBE
requests
<br>> >arrive from the same user. In REGISTER,
<br>> >&nbsp;&nbsp;&nbsp;&nbsp; they replace old ones if the SIP url is
same and additive otherwise.
<br>> >However, in SUBSCRIBE, I think it would make sense to make it additive
<br>> >only, since the same SIP url can subscribe to multiple events using&nbsp;
diff.
<br>> >subscription messages. Is this correct ?
<br>>
<br>> That sounds good, except possibly for the same "call leg". I think
the
<br>> part that needs to be clarified is: if a user sends a SUBSCRIBE
<br>> which has the same Call-ID, From, and To as a previous request,
<br>> does it add events or override them?&nbsp; I'm leaning towards overrriding
<br>> events, since there is no easy way to remove events otherwise. Note
well
<br>> that this wouldn't apply to other subscriptions with different Call-IDs,
<br>> which would (of course) remain additive.
<br>>
<br>> ARC> Somehow, I feel Call-ID and suscription should not be tied together.
<br>> This is primarily
<br>> ARC> because callid is generated by a terminal and a user has no
control
<br>> over it.
<br>> ARC> However, a subscription is for a user, not a terminal. so if
I
<br>> subscribe to a set of services
<br>> ARC> today, if I travel tomorrow, I want to walk into a Cyber Cafe,
log in
<br>> using some applet based
<br>> ARC> sip client that will generate a different Call-ID from my old
desktop
<br>> one an still be able to
<br>> ARC> access my old subscription list and add/del from it without
having to
<br>> know the Call-Id - which
<br>> ARC> in many cases, I never get to see as a user.
<br>> ARC> but the moment the backend notifier works on Call-ID - then
I would
<br>> never be able to 'Add' to
<br>> ARC> my list since Ill be using a different Call-ID in all probability,
<br>> neither will I be able to
<br>> ARC> somehow retrieve the old one, since match will never be true.
<br>> ARC> This is why I feel subscription should not be linked to a call
- id.
<br>>
<br>> (Incidentally, I like your example. I have recently had
<br>> discussions with several different groups of people who
<br>> wanted to know how one would go about, for example, having
<br>> a light blink on your phone and/or have a distinctive dial
<br>> tone when a message is waiting).
<br>>
<br>> ARC> Well, we actually had exactly the above requirement once, and
we did
<br>> it my adding
<br>> ARC> an Event like field with more or less the same semantics as
we
<br>> discussed above.
<br>>
<br>> >Event =&nbsp; 1#(eventid ; duration)
<br>> >event = token&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; // Does it need
to be broken ?
<br>> >duration = deltaseconds | SIP-Date
<br>>
<br>> I don't really like this. Beyond the problems with using
<br>> commas for separators *and* having commas appear in
<br>> the duration (if the SIP-Date format is used), I'm trying to
<br>> keep it as congruent as possible with REGISTER, since the
<br>> semantics for that are already well defined -- and code is
<br>> already written to handle it.&nbsp; For the above situation, I'd
<br>> propose that you either: a) send all three REGISTER messages,
<br>> with different Call-IDs, in a single UDP packet, or b) send
<br>> a single REGISTER for all three events with an expires of 2
<br>> hours. At the end of the 2 hours, re-SUBSCRIBE for just B
<br>> and C (expires in one hour), and one hour later, re-SUBSCRIBE
<br>> for just C. (A variation on this theme would be to subscribe
<br>> for 4 hours, override it in 2, and then again one hour later;
<br>> I prefer this method, since it eliminates the chances of missing
<br>> events -- although it counts on the addition vs. replacement
<br>> semantics I describe above).
<br>>
<br>> ARC> I assume where ever you mentioned REGISTER above , you meant
SUBSCRIBE
<br>> ARC> Again, solution 1 is to me, a kind of loose workaround. And
the other
<br>> solutions
<br>> ARC> force the client to maintain time differentiators actively since
it
<br>> must physically
<br>> ARC> maniuplate subscriptions to implement the common requirement
above.
<br>> Rather, The
<br>> ARC> Notifier is actually already implementing a timer list which
can
<br>> notify users
<br>> ARC> based on differential time values maybe, so why force the client
into
<br>> maintain the
<br>> ARC> differential list also ? To me it seems to be much cleaner for
the
<br>> client to specify
<br>> ARC> it once, and forget about it, as the notifier server is there
just to
<br>> remind him anyway
<br>> ARC> so why bog him down with keeping track of such requests ?
<br>> ARC> The only -ve point seems to be parsing for commas (which is
a non
<br>> issue for good parsers)
<br>> ARC> - but if that really is a problem, maybe we could work at a
simpler
<br>> expiry type ?
<br>> ARC> However, I dont feel parsing should be the reason to adopt the
<br>> alternate methods suggested
<br>> ARC> above. What do you/others feel abt it ?
<br>>
<br>> I must have been somewhat unclear. I am not proposing that one
<br>> use REGISTER itself to cancel a subscription; merely that one
<br>> use the same *mechanism*, with a SUBSCRIBE method. For
<br>> example, let's say that I want to cancel the subscription(s)
<br>> above. I could do so by sending:
<br>>
<br>> ARC> Sorry, I misinterpreted, then. Seems fine to me.
<p>--
<br>Jonathan D. Rosenberg&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
200 Executive Drive
<br>Chief Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Suite 120
<br>dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
West Orange, NJ 07052
<br>jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
FAX:&nbsp;&nbsp; (732) 741-4778
<br><a href="http://www.cs.columbia.edu/~jdrosen">http://www.cs.columbia.edu/~jdrosen</a>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
PHONE: (732) 741-7244
<br><a href="http://www.dynamicsoft.com">http://www.dynamicsoft.com</a></blockquote>

<pre>--&nbsp;
Manoj Bhatia&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;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |
manojb@cisco.com&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;&nbsp;&nbsp; :|:
7025 Kit Creek Road, P.O. Box 14987&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp; :|||||:&nbsp;&nbsp; :|||||:
Research Triangle Park, NC 27709&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp; .:|||||||:.:|||||||:.
919-392-3873&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; fax: 919-392-6801&nbsp;&nbsp;&nbsp; |&nbsp; C i s c o S y s t e m s</pre>
&nbsp;</html>

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Sunitha Kumar wrote:
>
> How do we match requests to responses at the proxies.( other than the
> branch in the Via);
> using preferably the fields in a CallLeg, ( so that the call can be
> uniquely identified).
>
> Ex: responses to 2 INVITES
> Is it legal for the proxy to tag the from field of every request that
> it forwards or proxies on, and remove the From tag, when the response
> to that comes back, before passing it  downstream.  Making sure, that
> each proxy on the way, adds itself to the record route.  So, they are
> traversed on the way back.

Why do this? You'll break authentication and it won't work if a tag was
already there.

>
> on other thoughts, what about incrementing the CSeq for every request
> that goes out, and decrementing the CSeq , when the response for that
> arrives, before passing it downstream.

I also highly recommend against this. If the client sends a request with
that same CSeq, which does not pass through your proxy somehow, things
will be mightily confused. Also this doesn't work with authentication.
Why do you want to do this? The spec is quite clear on how responses are
matched to requests. Why does incrementing the CSeq help in any way?

-Jonathan R.

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




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Steve Little wrote:
>
> Hi all,
> This Q is wrt construction of Route header by called
> EP. (6.32.3, bis)
> It mentions that while sending future requests, the
> called EP copies the Record-Route entries into is
> Route header preserving order.
>
> So, If say from A(Calling)->B(Called), a request
> from A took path A-P1-P2-P3-B, ie
> Record-Route: P3, P2, P1
> Then when B sends a BYE, it copies the Rec-Route as
> Route: P3, P2, P1, A
>
> Above makes sense. But what I dont understand is the
> statement that URIs contaned in Rec-Route are not
> useful in reverse path.

Thats because these URIs all refer to addresses for B, not for A.
Remember, the Record-Route header is built by copying the request URI of
the request as it goes from A to B. So, lets be specific about your
example:

B is actually userB@company.com. So, A sends an INVITE like this:

INVITE sip:userB@company.com SIP/2.0
Contact: sip:userA@university.edu
....

to its local proxy P1. P1 looks up company.com (P2) and forwards it
there, and adds itself to the Record-Route. The forwarded request looks
like:

INVITE sip:userB@company.com SIP/2.0
Contact: sip:userA@university.edu
Record-Route: sip:userB@company.com;maddr=<P1 address>


Now, company.com (P2) checks for userB in its database, and finds that
this is Joe.B@sales.company.com. So, it adds itself to the Record-Route
and forwards the request to sales.company.com (P3):

INVITE sip:Joe.B@sales.company.com SIP/2.0
Contact: sip:userA@university.ede
Record-Route: sip:userB@company.com;maddr=<P2 address>
Record-Route: sip:userB@company.com;maddr=<P1 address>

Now, P3 (sales.company.com) knows where Joe is right now because of
registrations. Its sip:Joe.B@host3.sales.company.com. So, it adds itself
to the record-route and forwards it there:

INVITE sip:Joe.B@host3.sales.company.com SIP/2.0
Contact: sip:userA@university.ede
Record-Route: sip:Joe.B@sales.company.com;maddr=<P3 address>
Record-Route: sip:userB@company.com;maddr=<P2 address>
Record-Route: sip:userB@company.com;maddr=<P1 address>

now, it arrives at B (Joe.B@host3.sales.company.com), and the call is
up. Now, Joe wants to send that BYE to hang up. If, as you suggest, it
constructs the Route header with the same URI, the request it sends will
to P3 look like:

BYE sip:Joe.B@sales.company.com SIP/2.0
Route: sip:userB@company.com;maddr=<P2 address>
Route: sip:userB@company.com;maddr=<P1 address>
Route: sip:userA@university.edu

Note that the request URI of this request is actually B, not A. This
request above will actually be routed correctly, so long as (1) every
proxy obeys the route headers, (2) every proxy uses the maddr parameter.
If any proxy messes either up, the request is probably going to loop
back to B. Thats bad. Thus, the mechanism is brittle. By having the
called party construct the BYE in the following way:

BYE sip:userA@university.edu SIP/2.0
Route: sip:userA@university.edu;maddr=<P2 address>
Route: sip:userA@university.edu;maddr=<P1 address>
Route: sip:userA@university.edu

We guarantee things work right, even if any proxy ignores the Routes or
ignores maddr. The request is still routed correctly. Thats why its done
this way. We wish to preserve the rule that the request URI indicates
where the request is going.



> In the above example, am I not re-using the URLs in
> the record-route for the reverse path ?
> Also, Im not clear  what the statement
> "However, all components except the maddr parameter
> are replaced by the calling URI." means - calling URI
> replaces what ? the Route list ? I guess not.

I think the example clarifies this. Only the maddr parameter from the
Record-Route header is used when the UAS constructs the Route header.

-Jonathan R.

--
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com




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Subject: [SIP] RE: TRIP for gateway registrations
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I would point out that H.323 RAS messages are not normally
used to register a range of phone numbers for gateways as it
is quite impractical, especially for gateways handling a large
number of phone numbers, and when keep-alive messages are used.

While some implementations uses tricks such as registering
"dialling prefixes" to map to telephone number, it is not the
intent of prefixes which are supposed to be used for special services.

Normally, a gateway registers an alias with its gatekeeper and the binding
between the alias and the range of phone numbers handled by the
gateway is "manually" configured in the gatekeeper, as described
in the draft. I would think that TRIP could automate this process.

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thursday, March 09, 2000 11:33 PM
> To: sip@lists.research.bell-labs.com;
> iptel@lists.research.bell-labs.com
> Subject: TRIP for gateway registrations
>
>
> Folks,
>
> Every once in a while, the question comes up of why SIP
> REGISTER is not
> the right vehicle for allowing gateways to register phone
> prefixes with
> a proxy. I believe TRIP is far better for this purpose. As it
> turns out,
> a small subset of TRIP is needed to solve this problem, and yet its
> still interoperable with full blown TRIP. Hussein Salama and I have
> written an I-D that discusses this. Until it appears in the archives,
> you can pick up a copy at:
>
> http://www.cs.columbia.edu/~jdrosen/papers/draft-rs-trip-gw-00.txt
>
> Thanks,
> Jonathan R.
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
>
>

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I think we are once again jumping the gun a bit. There are some much
bigger issues to address with a generic sip subscription/notification
mechanism:

1. what entities are generating the notifications? It can be a UA, or it
could be a proxy or registrar.

2. What are the types of events we want to support subscriptions for? I
know Adam has tried to keep that the subject of future extensions.
Problem is, this issue has a strong impact on the addressing, event
description and scaling capabilities. I don't think we can leave this to
future extensions without having the "base" version have nothing but two
methods. That doesn't really help much. I think we can define reasonably
broad classes of events. Certainly we aren't interested in supporting
events like "let me know when the coffee machine is done".

3. To whom does one address a subscription? If we address it to a user,
what does that mean? Are we asking for a subscription to that users
registration state?

Also, before we get to far with architecture and design issues, there is
the question about whether this concept, in general, is something the
working group is interested in pursuing.

-Jonathan R.



archow@hss.hns.com wrote:
>
> Hi,
> responses embedded and preceded by ARC>
>
> >The draft does not mention what happens when multiple SUBSCRIBE requests
> >arrive from the same user. In REGISTER,
> >     they replace old ones if the SIP url is same and additive
otherwise.
> >However, in SUBSCRIBE, I think it would make sense to make it additive
> >only, since the same SIP url can subscribe to multiple events using
diff.
> >subscription messages. Is this correct ?
>
> That sounds good, except possibly for the same "call leg". I think the
> part that needs to be clarified is: if a user sends a SUBSCRIBE
> which has the same Call-ID, From, and To as a previous request,
> does it add events or override them?  I'm leaning towards overrriding
> events, since there is no easy way to remove events otherwise. Note well
> that this wouldn't apply to other subscriptions with different Call-IDs,
> which would (of course) remain additive.
>
> ARC> Somehow, I feel Call-ID and suscription should not be tied together.
> This is primarily
> ARC> because callid is generated by a terminal and a user has no control
> over it.
> ARC> However, a subscription is for a user, not a terminal. so if I
> subscribe to a set of services
> ARC> today, if I travel tomorrow, I want to walk into a Cyber Cafe, log
in
> using some applet based
> ARC> sip client that will generate a different Call-ID from my old
desktop
> one an still be able to
> ARC> access my old subscription list and add/del from it without having
to
> know the Call-Id - which
> ARC> in many cases, I never get to see as a user.
> ARC> but the moment the backend notifier works on Call-ID - then I would
> never be able to 'Add' to
> ARC> my list since Ill be using a different Call-ID in all probability,
> neither will I be able to
> ARC> somehow retrieve the old one, since match will never be true.
> ARC> This is why I feel subscription should not be linked to a call - id.
>
> (Incidentally, I like your example. I have recently had
> discussions with several different groups of people who
> wanted to know how one would go about, for example, having
> a light blink on your phone and/or have a distinctive dial
> tone when a message is waiting).
>
> ARC> Well, we actually had exactly the above requirement once, and we did
> it my adding
> ARC> an Event like field with more or less the same semantics as we
> discussed above.
>
> >Event =  1#(eventid ; duration)
> >event = token       // Does it need to be broken ?
> >duration = deltaseconds | SIP-Date
>
> I don't really like this. Beyond the problems with using
> commas for separators *and* having commas appear in
> the duration (if the SIP-Date format is used), I'm trying to
> keep it as congruent as possible with REGISTER, since the
> semantics for that are already well defined -- and code is
> already written to handle it.  For the above situation, I'd
> propose that you either: a) send all three REGISTER messages,
> with different Call-IDs, in a single UDP packet, or b) send
> a single REGISTER for all three events with an expires of 2
> hours. At the end of the 2 hours, re-SUBSCRIBE for just B
> and C (expires in one hour), and one hour later, re-SUBSCRIBE
> for just C. (A variation on this theme would be to subscribe
> for 4 hours, override it in 2, and then again one hour later;
> I prefer this method, since it eliminates the chances of missing
> events -- although it counts on the addition vs. replacement
> semantics I describe above).
>
> ARC> I assume where ever you mentioned REGISTER above , you meant
SUBSCRIBE
> ARC> Again, solution 1 is to me, a kind of loose workaround. And the
other
> solutions
> ARC> force the client to maintain time differentiators actively since it
> must physically
> ARC> maniuplate subscriptions to implement the common requirement above.
> Rather, The
> ARC> Notifier is actually already implementing a timer list which can
> notify users
> ARC> based on differential time values maybe, so why force the client
into
> maintain the
> ARC> differential list also ? To me it seems to be much cleaner for the
> client to specify
> ARC> it once, and forget about it, as the notifier server is there just
to
> remind him anyway
> ARC> so why bog him down with keeping track of such requests ?
> ARC> The only -ve point seems to be parsing for commas (which is a non
> issue for good parsers)
> ARC> - but if that really is a problem, maybe we could work at a simpler
> expiry type ?
> ARC> However, I dont feel parsing should be the reason to adopt the
> alternate methods suggested
> ARC> above. What do you/others feel abt it ?
>
> I must have been somewhat unclear. I am not proposing that one
> use REGISTER itself to cancel a subscription; merely that one
> use the same *mechanism*, with a SUBSCRIBE method. For
> example, let's say that I want to cancel the subscription(s)
> above. I could do so by sending:
>
> ARC> Sorry, I misinterpreted, then. Seems fine to me.

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






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Subject: [SIP] New I-D available soon - Java enhanced SIP
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Hi,

I've posted an Internet Draft to the internet draft archives which
describes
an extension to SIP to build on some of the strengths of Java, the JVM
concept and Java Mobile agents.

It is not intended to discuss this at Adelaide as it is an initial draft
but
I would be interested in getting feedback and comments from anybody who
takes a look at it.

The ID defines the extension to the SIP protocol and describes how it is
intended to:

- extend SIP messages to carry Java applets or Java applets plus their
state
and runtime contexts - in other words  Java mobile agents (or URLs to
either
the applet or the Java Mobile agent)

- to define a Java SIP API which will allow the Java applet or Java Mobile
Agent interact with the receiving host system and receiving SIP client

- to extend the SIP client so that the Java applet or Java Mobile Agent is
run before any other actions are taken on the receipt of a message by the
receiving host SIP client.



It also includes security measures to allow a client to be configured to
enable and disable certain actions between the JVM running the Java applet
or the Java mobile agent virtual machine running the Java Mobile Agent in
the message and the clients SIP client and host system.

Cheers,

Mick O'Doherty
Nortel Networks

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

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


     Title          : Java enhanced SIP (JES)
     Author(s) : M. O'Doherty
     Filename  : draft-odoherty-sip-java-enhanced-00.txt
     Pages          : 9
     Date      : 08-Mar-00

This document defines an extension to the SIP [2] protocol to do a
number of things - to extend SIP messages to carry Java applets or
Java Mobile Agents, to define a Java SIP API which will allow the
Java applet or Java Mobile Agent to interact with the receiving
host system and to extend the SIP protocol client so that the Java
applet or Java Mobile Agent is run before any other actions are
taken on the receipt of a message by the receiving host SIP
client.

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

Internet-Drafts are also available by anonymous FTP. Login with the
username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
     "get draft-odoherty-sip-java-enhanced-00.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
----------------------------------------------------------------------------

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

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To: Igor Slepchin <islepchin@dynamicsoft.com>
Cc: Sunitha Kumar <skumar@vovida.com>,
        SIPbell-labs <sip@lists.research.bell-labs.com>
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Let me make a general comment here on mailing list usage.

The SIP mailing list is an excellent place to address questions
regarding the specification. However, the purpose of the list is to work
on our deliverables as a working group. Time spent by the group
answering questions that have been discussed before is less time the
group can spend making forward progress on our documents. So, to try and
keep focused, I would encourage people to do the following. Before
asking your question to the list, please check the FAQ, the archives,
and the rfc2543bis draft to see if your question is answered there. If
not, then of course go ahead and ask on the list.

Thanks,
Jonathan R.

Igor Slepchin wrote:
>
> It certainly should, and not only for BYE requests. This is discussed in
> http://www.cs.columbia.edu/~hgs/sip/notes.html (see the note on
> Record-Route towards the end of the document) and in
> http://www.cs.columbia.edu/~hgs/sip/drafts/draft-ietf-sip-2543bis-00.pdf
> (section 6.3.33).
>
> ---
> Igor Slepchin
>
>
> Sunitha Kumar wrote:
> >
> > It is given that the record route in 200 OK is copied into  the Route
> > headers for subsequent requests from the caller.  My question was, if
> > we needed to make the BYE from the *callee*  traverse the same set of
> > proxies, then  should this be rememberd from the INVITE request
> > obtained from the caller, or does the via field help in that.
> >
> > Thanks!
> >
> > --
> > Sunitha Kumar
> > Software Engineer
> > Vovida Networks
> > (408) 957 - 6374
> >
> >

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




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

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

-Jonathan R.

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

---------
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Anders Kristensen wrote:
>
> The draft talks about when it's OK to generate new reliable responses
> for a transaction, but it doesn't seem to say anything about whether
> final responses can be generated by a UAS which has outstanding
> unacknowledged reliable responses.
>
> I believe the UAS must be allowed to generate a final response at any
> time.

Yes, agreed.

>
> The question is what happens with existing 'orphan' reliable responses
> and corresponding PRACKS. They can either terminate normally, i.e. the
> UAS continue retransmitting those reliable responses as it would have
> done had a final response not been sent and likewise the UAC still
> PRACKS them, or else maybe they could be terminated somehow - the UAS
> stops retransmitting them, and the UAC never PRACKS anything for which
> it has received a final response.

Hmm. Good question. I tend to lean towards having those orphaned
provisional responses still be PRACKED. Here's my reasoning:

1. These responses were sent reliably since they contain some important
and useful information. It may be that this information is no longer
useful once the final response is sent, but then again, it may be that
it is still useful. So, if the user tries to send a provisional response
reliably, lets always deliver it. Let the application decide whether its
useful after a final response.

2. Lets keep transactions independent. We went through a similar thing
with CANCEL, and the state machines were confusing until we decided to
make CANCEL a mini-transaction that does nothing but ask the UAS to
quickly answer the request. I think we will see confusion in the PRACK
processing if its operation becomes dependent on events in other
transactions.

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

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




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Subject: [SIP] question on sip url grammar.
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A question on BNF grammar in SIP url.
In BNF RFC, it is given that [       stands for 0 or 1 times the value.
It is given

SIP-URL = "sip:"  [userinfo '@'] hostport url-parameters....
user-info = user [":"password]
user = *(unreserved... )                          //optional

password = *(unreserved ...)                 //optional.

so, this leads to a url as:

sip::@hostport.                  So, with the above grammar, this is
valid.
is my understanding right?

Also, for hostport;

sip:userinfo@host:urlparms            //since port is optional.

Thanks much!

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



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Subject: [SIP] Revised Agenda for SIP WG at IETF 47
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Ok, I asked for glaring errors in our first agenda draft, and YOU PEOPLE
FOUND SOME! Darn.

So, I've revised the agenda and posted it anew at:

http://www.softarmor.com/sipwg/meets/IETF47/agenda.html

and attached an HTML version to this email.

Note that we are getting a rather seriously packed agenda. I propose that
we
settle further scheduling conflicts with noble denial-of-service duels at
dawn. Two kernels, one hub, pings of death . . .

--
Dean Willis
SIP WG co-chair

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From sip-admin@lists.bell-labs.com  Mon May  8 20:56:40 2000
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          --> Kimmo.Rantanen@lmf.ericsson.se writes:
          >I like to add one more question:
          >Is it absolutely necessary that both direction use the same
type of
          >codec ?

          No.

          This is rather undesirable for vocoders or implementations
employing advanced echo cancellation, intelligent jitter buffer management
or a host of other DSP or voice-related properties.

          From my reading of the RFC, it is also equally possible for
either side to chop-and-change between vocoders at any point in time
(without SIP message exchange). Supporting this mixing and splicing of
vocoders significantly complicates the playback engine - the ability will
most likely be unsupported on simpler implementations or at best may cause
small audio glitches.

          It would be nice if the caller could force the callee to
agree on a single common vocoder if it doesn't support the use of multiple
vocoders. [Making the SDP session closer to a true two-way media exchange
rather than simply two disjoint vocoder packet flows in each direction.]

          One approach might be to place an updated SDP body in the
ACK where the SDP body has only one media format selected. The RFC
currently
prohibits changing the SDP body in the ACK (unless the INVITE had no SDP
body). Does changing the body introduce any major problems or
inconsistencies ?

          Furthermore, IMO it would be desirable if the callee could
indicate their vocoder preference by ordering the list of RTP media format
values in the 183/200 response.

          How have other people tackled this problem ? Are there any
drafts or FAQ on these subjects.

          Cheers,

          Robert.





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Rohan Mahy wrote:
>
> Section 4.2.3 (The OPTIONS Method) goes on to say that user agents MAY
> respond to OPTIONS.  I would infer from this (and a quick skim of Section
> 12) that there is no requirement for a proxy to respond to a method it
> doesn't understand.
>
> Is there still any compelling reason to require the 200 OK after PRACK?

Well, perhaps you should read the rest of the sentence. There are words
after "MAY respond to this request". Lets be specific here. This section
says:

> The server is being queried as to its capabilities. A server that
>    believes it can contact the user, such as a user agent where the user
>    is logged in and has been recently active, MAY respond to this
>    request with a capability set. A called user agent MAY return a
>    status reflecting how it would have responded to an invitation, e.g.,
>
>
>
>
> Handley, et al.             Standards Track                    [Page 29]
>
> RFC 2543            SIP: Session Initiation Protocol          March 1999
>
>
>    600 (Busy). Such a server SHOULD return an Allow header field
>    indicating the methods that it supports. Proxy and redirect servers
>    simply forward the request without indicating their capabilities.
>
>    This method MUST be supported by SIP proxy, redirect and user agent
>    servers, registrars and clients.


That is it MAY respond with a capability set, or MAY respond with a
status reflecting how it would have responded to the invitation. The MAY
is a choice among how the response is formulated, NOT on whether a
response is sent.

All requests are responded to. period.

By not having the 200 OK to PRACK, we break the transactional model of
the protocol. This means proxies and UAs have all this extra state
laying around. It means spurious CANCELs and responses might get sent.
It means we probably need a Proxy-Require, and I really don't want to do
that, because it means compatibility is broken.

-Jonathan R.



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




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Adam wrote:
> Dean blurted
> >Something like
> >
> >  INVITE premiumservice@gateway.carrier.com
> >  To: tel:5551212;trans=lnp?gdbs
> >
> >could be used to indicate to gateway.carrier.com that premium gateway
> >service is requested towards the PSTN location identified by telephone
> >number 5551212, that LNP and GDBS translation have been applied, and
that
> >the phone number is of local scope for the gateway.
>
> Since the To: field isn't allowed to change during the transaction,
> aren't you going to run into problems if some service along the way
> determines that the called number needs to change (as the result of
> LNP, 800/900 lookup, etc)?

Right you are. That should teach me not to come up with examples while
squatting on one of those interminable telco conference calls.

But I still think using a tel: url gives a cleaner semantic than
sip:number@gateway;user=phone.

I was talking with some guys today about scope and context, and I'm
inclined
to think we need the nature of address and numbering plan indicator fields
out of some ISUP variants. Scope appears to be global, local, or private,
with some need to represent local and private within some context like area
code or private numbering plan.

So, for example

tel:+19727294162 is a global number with unspecified context,
tel:9727294162 is a local number
tel:7724162;phone-context=mcivnet34 is a private number with context in the
MCI private numbering plan #34
tel:4512345687724162;phone-context=mciswtrunk is a private number, in
numbering plan 45, relative to switch 123 and trunk group 45678
. . . and whatever other sick private variants one might come up with


--
Dean


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Mark Foster wrote:
> That at least some forms of Iptel must comply with the non-reliance
requirement in at least some countries suggests that the
> SIP, TRIP, enum, etc., must be flexible enough to allow implementations
that will satisfy these requirements, and still be
> interoperable with those that don't.
>
> In the US and Canada, the key to this was the N-1 trigger placement
policy.  This was simply a guideline on which networks
> should be prepared to dip calls placed to which areas in order to not
force calls to be re-routed at the donor network
> (termination routing).  Key to having flexibility on where triggers are
placed, resulting in NPDB queries, is having a form of
> post-query signaling that makes the query results available to all
downstream networks.

I'm not really sure that the N-1 concept works that well in an IP
environment:

1. If you look at DNS as a similar service to LNP on the IP side, the
originating host does a name to address translation right away. This
happens even if the name thats being translated is "far away". This
avoids triangle routing, as has been discussed.

2. The interconnection of SIP servers and providers within the IP world
is not that rigid. I'm not sure how a SIP proxy would indeed determine
that it is the N-1th server in the chain.

In any case, determination of where the NPDB query is done is something
I don't think is within the scope of iptel to determine. All we need to
figure out is what kind of numbers are distributed in TRIP. Near as I
can tell, since LRNs still refer to the prefix of numbers served by a
switch, doing what we were doing all along works just fine.

>
> The key data elements in ANSI ISUP are: the LNP-query-done indicator (bit
in the FCI); placing a routing number returned
> from the NPDB in the CdPN (LRN or whatever), and moving the original
dialed number to a GAP.  Subsequent call routing is
> done on the routing number, while not loosing the dialed portable
subscriber number.  Given the need for SIP to interoperate
> with ANSI ISUP, there should be some consideration of implementing these
semantics.
>
> If SIP captured these fields, so as to provide a post-dip form of
downstream signaling, and TRIP were designed to build route
> tables based on administratively assigned number ranges (which, by
definition would make calls to routing numbers [LRN]
> routable) , and enum did not require DNS queries to be answered by donor
networks, then the existing regulatory
> requirements for LNP could be met.

First off, SIP has two fields that identify the destination party. The
To field represents the original number that the caller called, and the
request URI represents the address translated at each hop. Thus, the To
field carries the original dialed number, and the request URI would
carrying the routing number. Now, do we need to somehow indicate that
the LNP dip has been done? I'm still unsure about that. I believe
Henning had suggested that the use of a sip URL (as opposed to a tel
URL) in the request URI would effectively be an indicator of such. That
aside, I think there is a possible confusion if one were to assume an
LRN looked like a normal SIP URL:

INVITE sip:1212344 SIP/2.0
To: sip:14081234567

The problem is related to past discussions on the SIP mailing list
regarding overlap dialing. A terminating gateway would look at
sip:1212344, and know how to route it, but might not know whether this
was a dialed number with more digits coming, or an LRN. Does it matter?
Not being an ISUP expert, I'm not sure. If it does matter, something
like user=lrn within the SIP URL would fix that.

But anyway, thats also not a problem for iptel, although I've cc'd the
sip list on this message.

-Jonathan R.

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

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Another option is just a SIP MIME bodypart. No binary-to-text
translation needed there. Would be nice to spec this topic out.
--
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs




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Robert Sparks wrote:
>
> > Why? I really liked the idea of knowing that this call was
> > the result of
> > a transfer, and knowing who the transferor was. A UA can always ignore
> > this information, and then get the basic service you have today.
>
> I really like the idea too. Perhaps as part of the work right now we
> can capture knowing that the call was the result of a transfer (by adding
> an informational header to the invite - something that doesn't change
> the invite semantics), but my preference is to wait for direction from
> the SIP security task folks before addressing telling the transfer target
> who the transferor was.

There isn't going to be any magic poof that causes a solution to be
generated. I think its the job of the author of the extension to propose
some security mechanism so that there can be a starting point for
discussion.

>
> > The document says TRANSFER can't have a body. Why not? I can think of
> > some cool uses for this down the road. Why rule it out?
>
> For consistency of implementation? (btw - One of those cool uses could be
> providing the identity of the transferor.) The presence of a body
> would likely affect the semantics of the operation (if it were honored,
> hence the consistency comment) through side-effects if not directly,
> so would it not make sense for the addition of a body to the base
> method be an actual extention?

The presence of a body should not, in any way, alter the basic semantics
of the request. Treatment of a body in TRANSFER is no different than
bodies in any other request. If the UAS doesn't know what to do with it,
it sends a 415. If it does know what to do with  it, it gets processed.


> > This thing about the Call-ID being copied from the TRANSFER
> > seemed odd.
> > Why is that? If you wanted to accomplish this, it seems better to have
> > the URL in the Transfer-To header contain the Call-ID as a URL
> > parameter. In other words, why not do:
> >
> > Transfer-To: sip:user@host?Call-ID=9nasd09asd--asdasd98ays
>
> Interesing thought! Can we explore your thinking more here?
> Using this, the TRANSFER would look like
>
> TRANSFER sip:transferee@transfereehost
> Transfer-To: sip:transfer-target@targethost?Call-ID=abc@eeOrorhost
> Call-ID: abc@eeOrorhost
>
> Are you suggesting the subsequent invite look like this?
>
> INVITE sip:transfer-target@targethost?Call-ID=abc@eeOrorhost
> Call-ID: def@eehost

No, like this:

INVITE sip:transfer-target@targethost
Call-ID: abc@ee0rorhost

This way, the tranferror could ask the transferred party to add other
headers or even body elements.


> On reflection, I specified this copy because it resulted in
> a call-leg tuple that looked the same as it would have after
> a sip-cc-01 blind transfer. The motivation that draft had for
> this behavior may not apply here. Is there any value in keeping
> the Call-ID information accross this function (thinking of it
> as PLEASEINVITE instead of TRANSFER)?

Perhaps. By allowing the Call-ID to be used to be placed in the URL in
the Transfer-To header, the transferror can make the call-id the same,
different, or unspecified. This seems to result in maximum flexibility.


Later, Henning Schulzrinne writes:
> >
> > It would be really nice, however, to know ahead of time whether all
> > parties supported transfer (and any other mechanisms to be defined).
The
> > Supported extension draft defines a way in which the UAC would indicate
> > to the UAS that it knows the new mechanism, but that draft does not
> > mandate that a UAS return a Supported header in the response to any
> > method (only OPTIONS). We can add this, and say that if a UAS receives
a
> > request (with any method) with a Supported header, it should place a
> > Supported header in the response and list those features it knows. This
> > would allow a caller to know whether the called party supports transfer
> > before trying it. Thoughts?
>
> It might make sense to have both Allow and Supported in responses other
> than OPTIONS. In this case, Allow seems the more appropriate solution,
> given that it's a new method and that having two mechanisms to ask for
> the same thing seems a bad design.

You're right, its Allow here, not Supported.

> Thus, suggesting that final responses
> include Allow doesn't break anything (a UAC could have a simple logic
> that did "if response didn't have Allow, send an OPTIONS just in case"),
> but it does cut down on traffic. It's much nicer if the UAC can gray out
> a Transfer button rather than pop up a failure note after the caller has
> said "let me transfer you...".

Definitely.

-Jonathan R.


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




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To: Anoop Tripathi <Anoop_Tripathi@mw.3com.com>
Cc: sip@lists.research.bell-labs.com
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Subject: [SIP] Re: Why must a stateless proxy insert a Via header for itself.
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Anoop Tripathi wrote:
>
> Currently, the RFC states that every server in the path of a request must
add
> itself as a Via header.
>
> As a stateless proxy, if this restriction is removed then I can process
more
> calls with faster call setup times.
>
> So , can we make inserting itself as a Via optional for a stateless
proxy.
>
> Thanks,
>
> Anoop

Then how do you do loop detection ??

--
Best regards,
Shail




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To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
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|-----Original Message-----
|From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
|
|> John Mainwaring wrote:
|>
|> |The problem is related to past discussions on the SIP mailing list
|> |regarding overlap dialing. A terminating gateway would look at
|> |sip:1212344, and know how to route it, but might not know whether
|> this
|> |was a dialed number with more digits coming, or an LRN. Does it
|> matter?
|> |Not being an ISUP expert, I'm not sure. If it does matter, something
|> |like user=lrn within the SIP URL would fix that.
|>
|> user=lrn is almost right.  If you have queried a portable number and
|> found that it is not ported, then the DN has the same status
|as an LRN
|> would have if the number were ported (i.e. it should be treated as
|> authoritative for routing).  user=np-queried might be closer to the
|> mark.
|
|I wonder if we can't think about this problem in more general terms.
|
|In the SIP model, each proxy owns some piece of the namespace
|(engineering.dynamicsoft.com, for example), and is therefore
|responsible
|for the definition of the user space (jdrosen@) within that domain.
|Thats why each proxy can, an usually will, perform some kind of URI
|translation independent from other proxies. Our underlying assumption
|was that usernames only have meaning within their domain. For example,
|with sip:jdrosen@dynamicsoft.com, only a dynamicsoft.com proxy
|knows how
|to handle jdrosen. There might be jdrosen's in other domains
|that aren't
|me. However, in the case of telephone numbers, the proxy that owns the
|domain does not necessarily own the user space.
|sip:5551212@gateway.com,
|for example, uses a user name that is not "owned" by
|gateway.com, in the
|sense that this name has valid semantics outside of gateway.com. Other
|domains, like wcom.com, would know how to handle this number as well.
|Its possible that we might even see things besides phone
|numbers as user
|names that are valid across multiple domains. An example is (heaven
|forbid), social security numbers (in the US, at least):
|
|sip:123-45-6789@wcom.com;user=ssn
|
|Now, I might imagine that some kind of mobility services might be
|supplied for these names, so they might need to be translated also.
|
|I'll call these type of user names "cross domain user names".
|
|With cross domain user names, this is where we run into issues
|like "has
|the number been queried yet". However, I think we might see other
|translation mechanisms besides LNP. Soooo, might it not be useful to
|define URI parameters that indicate what global translation services
|have been applied to a cross domain user name?
|
|As an example:
|
|sip:15551212@cisco.com;user=phone;trans="lnp,gdbs"
|
|where lnp and gdbs are some kind of global translation services that
|apply to names of the type phone.
|
|For cross domain user names, we will always need a URI parameter to
|indicate the global namespace from which these names are drawn. In the
|case of phone numbers, thats user=phone. Our original intent with the
|user=phone parameter was to specify only the type of BNF used for the
|user part of the URI. But, the more general operation of
|specifying both
|the format and the namespace seems useful.

I think the problem is a bit different.  919-555-1212 uniquely identifies a
user within the North American Numbering Plan.  It's just that with LNP it
can no longer locate the user.  If someone wants to use
919-555-1212@vanity.org as an email address and someone else uses
919-555-1212@workplace.com, that's fine, but has nothing to do with
telephone numbers.  I think I want to be able to place a call when all I
know is 919-555-1212.

Without LNP, it would have been fairly easy to  discover that 919-555
implied bellsouth, so the number could have been given as
919-555-1212@bellsouth.com.  Now it is possible that some numbers from
919-555 might be ported to othertel.com.  Othertel might the native owner
of
919-444-xxxx numbers and might choose to host 919-555-1313 on their 919-444
exchange.  In that case, the LRN for 919-555-1313 would be 919-444.  Once I
have done an LNP query, I might want to use something like
919-555-1212@919-555.bellsouth.com for a number that has not been ported
and
919-555-1313@919-444.othertel.com for a number that has been ported to
othertel.

I can't shorten the 10 digit number in the user name, because of things
like
NPA overlays and switches that host multiple office codes.  I need the full
address to select the right user as well as the LRN to route to the right
place.

If a SIP user wants to call someone who uses a PSTN phone, the system needs
to deal with certain inconvenient aspects of the PSTN. A notation like
919-555-1212@919-555 or 919-555-1313@919-444 might be useful to indicate
that an LNP query has been done, and that authoritative routing information
is now available.  Appending text-based names such as
919-555-1212@919-555@onetel.com or 919-555-1313@919-444@anothertel.com
might
even allow subsequent routing to take place without further concessions to
PSTN quirkiness.

Of course this system could also be expanded to mobility services; routing
for 919-555-1313 could be set to 212-666-1414@212-666.bigappletel.com in
the
daytime and 919-555-1313@919-444.othertel.com in the evening if you went in
for that kind of commuting.

|> It's also necessary to define what the TRIP should do differently
|> depending on whether the np-queried indication is present.  I must
|> admit I came into this discussion mainly because I have worked on
|> LNP.  I assume the objective is not to route unqueried calls to the
|> donor proxy.
|
|Hmm. This is something thats also different from the PSTN. The cost of
|routing to the donor proxy in SIP is much less, since proxies can be
|stateless. Thus, if a proxy receives a call for some number that is
|ported, it stateless proxies the call towards the next destination. The
|result is that there is no triangle routing for media (there generally
|isn't), nor for subsequent signaling. There is only marginal cost for
|the donor provider. There are obviously polticial and business issues;
|one may not want to trust or rely upon the donor proxy at all,
|as is the
|case of LNP in the US. But, the general SIP model is that we assume
|proxies do route calls; no security measures can prevent a proxy from
|dropping messages. Why, in the case of LNP, do we now need to assume
|these proxies won't route calls?
|
|
|
|
| I have noted that originator query may not be practical
|> for nationwide calling and that donor proxy is too late unless you're
|> prepared to consider RTP.
|
|The assumption of the impractability of originator queries has
|everything to do with the existing structure of LNP. Its
|mainly because,
|AFAIK, the systems are different all over the world. However, if we use
|the IP model, with a DNS solution, its no harder to query for a number
|on the other side of the planet as it would be for a number next door.

No, it's not quite as simple as that.  The system is designed around the
strengths and weaknesses of the PSTN.  The the architecture of the pstn
makes it easy to route calls on partial information, but switches in the
pstn perform better if they do as few database queries as possible.
Switches work fine if they have no idea of the portability status of most
numbers; they are only set up to query nearby numbers.  A switch doesn't
need authoritative data that a number is not portable.  It just needs to be
set up to query that (local) subset of portable numbers which it would
misroute if it didn't query them.

The LNP databases are distributed because of the geographic restrictions on
Bell Operating Companies.  Each RBOC only has to worry about its own data,
and only has to route queries within its own SS7 signalling network.

There are actually two levels of database, the SCPs that actually handle
queries and the NPAC (?) which processes administrative change requests and
downloads the SCPs.  There could be several SCPs owned by different
operating companies that contain the same data; each operating company
would
route queries to its own SCP.  The NPAC service is provided by NeuStar,
formerly Lockheed Martin, and Mark Foster of Neustar has given us some
excellent information on LNP within this thread.

It seems unlikely that IP telephony would use the existing SCPs for
queries,
or that it would be based on SS7 TCAP messaging.  I don't have a clear idea
of what databases there would be for IP telephony, but they would certainly
have to arrange to download data from the NPAC.  There are costs associated
with the databases and with keeping the LNP information up to date.

If you want to do source routing, it will certainly take a non-trivial
effort to make North American or even world wide LNP information
accessible.
There would be a case for aggregating it with whatever centralized routing
servers there are, or for arranging for the destination's proxy to return
information about ported numbers instead of simply refusing the invitation.

|Also, pardon my ignorance, but what is RTP? Many of us on this
|list know
|RTP as the Real Time Transport Protocol (rfc1889) but I don't think
|thats what you mean.

In that context, RTP = Release To Pivot; switch A routes a call to switch B
which may provide some service and then releases the call back to A (the
pivot switch) with instructions to reroute the call to destination C.
Sorry, I thought I had expanded it earlier.

Is there any TLA in our businsess that only has one expansion?  What does
TLA mean apart from Three Letter Acronym?

| One answer might be for the TRIP to provide
|> a response on unqueried numbers that routes to an intermediate server
|> that does know how to do NP queries for the destination DN.
|
|TRIP is not a query protocol. Its a routing protocol. This would be a
|SIP function, and is already possible. "408 Not Found".

But why was it not found?  In the telephony e.164 numbering domain prior to
LNP, "not found" meant "does not exist".  LNP required switches to add
"might be somewhere else" logic.  It would be desirable for a routing agent
to say something more informative than "not found", if  it had any
additional information.

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To: "Liess, Laura" <Laura.Liess@telekom.de>
Cc: sip@lists.research.bell-labs.com, iptel@lists.research.bell-labs.com
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Subject: [SIP] Re: uploading services from the user's  home network to a sip
 server of  the visited network within a 3G.IP environement
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"Liess, Laura" wrote:
>
> Hi,
>
> after doing a very good job by deciding to take SIP for call control for
multimedia services within 3G.IP >networks, the 3G.IP operators have a lot
of open items concerning the architecture, one of them being which >network
controls the services for a roaming user: the home network, the visited
network or shared control. >The concrete question: how can I upload at the
begin or during the call the roamimg user's services >available in the home
network into the SIP proxy of the visited network? As far as I know, the
CPL and >REGISTER helps me to upload services only from the UA to the
proxy.

I'm not an expert in wireless, so I'm not certain of the scope of the
services we are talking about. Mid call services, for example, are not
currently within the scope of CPL. However, there is nothing which
prevents the CPL-upload-in-register mechanism from being used by a
server to upload scripts to another server. Registrations, in general,
can be used for third party controls. The From field identifies the
party initiating the registration.

-Jonathan R.

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




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        iptel@lists.research.bell-labs.com, james.yu@neustar.com,
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>There was some discussion of this problem wrt DCS during the November
>interim meeting. It was suggested that a cross-domain destination which is
a
>phone number could be better represented by a tel: URL than by confusing
the
>scope of the user part of a sip: URL. This simplilifies life greatly.
>
>Something like
>
>    INVITE premiumservice@gateway.carrier.com
>    To: tel:5551212;trans=lnp?gdbs
>
>could be used to indicate to gateway.carrier.com that premium gateway
>service is requested towards the PSTN location identified by telephone
>number 5551212, that LNP and GDBS translation have been applied, and that
>the phone number is of local scope for the gateway.

Since the To: field isn't allowed to change during the transaction,
aren't you going to run into problems if some service along the way
determines that the called number needs to change (as the result of
LNP, 800/900 lookup, etc)?

For example; let's say that one of the services that I have out in the
network is a "reachable from anywhere" speed dialing list. It maps,
for example, "4" as a telephone number to "2145551212". So I launch
an invite to my local proxy:

  INVITE sip:4@proxy.spleen.com SIP/2.0
  To: tel:4

It goes through my service node:

  INVITE sip:4@usersvc-34987.spleen.com SIP/2.0
  To: tel:4

And he wants to send it on to your gateway for premium service, after
translating the 4 into "2145551212."

It's my interpretation that, under the model you're proposing,
I'd need to send along

  INVITE sip:premiumservice@gateway.carrier.com SIP/2.0
  To: tel:2145551212

But that would break the transaction, since the To: field has changed.

I understand the distinction between destination and service
activation that you're trying to make, but your example seems
unworkable. What am I missing?

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

---------
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Anoop Tripathi wrote:
>
> Currently, the RFC states that every server in the path of a request must
add
> itself as a Via header.
>
> As a stateless proxy, if this restriction is removed then I can process
more
> calls with faster call setup times.
>
> So , can we make inserting itself as a Via optional for a stateless
proxy.
>
> Thanks,
>
> Anoop

Then how do you do loop detection ??

--
Best regards,
Shail




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David,

David Oran wrote:

> >The question is: is it posible to define, to any extent, a mechanism
> capable >of preventing the "hacked" terminal to bypass the proxy and
> call
> the >"standard" endpoints witout being billed ?
>
> There's no way to prevent one "hacked terminal" from talking to
> another
> "hacked terminal" directly, bypassing any proxy.
>

Fine, this is just impossible to prevent.

Then, I had in mind some via-header based solutions but, eventually, I
managed to hack all of them. It looks like the one you propose requires
some more thinking.....

> You can, at some pain, protect against is having a "standard terminal"
>
> answer a call from a "hacked terminal". The way you do this is when a
> SIP
> message comes into the UAC, ou look at the last VIA on the via list.
> You
> then get the IP address of that machine (either because it was on the
> VIA
> istelf or you called DNS to get its address) and compare that with the
>
> source address of the SIP message.

I assume you mean "Source address of the IP packet containing the SIP
message", right ?

> If they don't match, you know somebody is
> trying to spoof you.

mmm, let me think....

got it !?!?!?

   * 1) what if there' s a NAT in between ?

   * 2) vhat if the proxy "must be in the list", but not necessarily as
     the last one ? (E.g. because the proxy that bills you is that of
     your "home network", and you have moved somewhere else)

   * 2) on the other hand, if you  want the billing proxy always to be
     the last one (apart from all the clearinghouse-related issues Henry
     was mentioning today), it must be communicated upon registration. I
     must think about this, but I have the feeling that a possibility
     might be that  I can fool the "standard" terminal to re-register
     with me (the hacked terminal) making him believe I am "the proxy".
     I would obviously be adding my address to the VIA header when
     inviting him.

   * 3) I will think a little bit more about it

   * 4) Anyway, do yo agree with me that the complexity of the "crack"
     required to "patch" such features on the "standard" terminal (going
     back therefore to the first case you mentioned) is definetly much
     smaller than that of installing a complete protocol from scratch on
     a device that was not designed to run it ?

Thank you very much and
Best Regards
Giuseppe

>
> If they DO match, then if the SIP UAC knows the list of SIP proxies
> which it
> trusts to deliver calls (and bill for them), the UAC either accepts or
>
> rejects the request appropriately.
>
> > Please CC replies to gricagni@tin.it (it is easier for me to read
> email
> from > such mailbox here in Adelaide)
>
> >Please let me know
> >Best Regards
> >Giuseppe

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Anup,

Sorry for my poor explanation.  From a global perspective, there will be
one
logical session.  This logical session will consist of a SIP session and a
RTSP session which need to be united.

Given this info, your option a) would not be appropriate.

option b) is what we'd like to do.

     >If you go with the model that the RTSP server understands SIP, then
RTSP
     >provides a means to pass the conference ID to the RTSP server so
that it can
     >join the SIP session and link subsequent RTSP operations to the SIP
session.

Wow, thanks for pointing this out.  Should've noticed this before.

The text is the RFC is quite clear:

   Separation of stream control and conference initiation:
          Stream control is divorced from inviting a media server to a
          conference. The only requirement is that the conference
          initiation protocol either provides or can be used to create a
          unique conference identifier. In particular, SIP [12] or H.323
          [13] may be used to invite a server to a conference.

====

I believe my problem is solved.  Thanks!



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

> -----Original Message-----
> From:   Anup Rao [SMTP:anrao@cisco.com]
> Sent:   Wednesday, March 15, 2000 4:23 PM
> To:     Linden deCarmo
> Cc:     'Kundan Singh'; 'Dean Willis'; sip@lists.research.bell-labs.com
> Subject:     Re: SIP & RTSP interworking?
>
>
>
> Linden deCarmo wrote:
>
> > I like this a lot better, but the problem I have is this, we'd like for
> BOTH
> > sessions to be active at the same time and both of your proposals
> require a
> > teardown of the SIP session.
> >
>
> Linden,
>
> I still dont understand what the application is. Your first note said :
>
> *************
> What I'd like to do is setup a session with SIP (via an INVITE), then
> control streaming aspects via RTSP (i.e. play, pause, stop etc.).
> *************
>
> Implying to me that there was only one session.
>
> Your second note says:
>
> **************
> I like this a lot better, but the problem I have is this, we'd
> like for BOTH
> sessions to be active at the same time and both of your proposals
> require a
> teardown of the SIP session.
> **************
>
> Implying that there were two. (which I think makes sense).
>
> I think there are broadly two scenarios:
>
> a)  Sending an RTSP URL to a SIP UA as part of redirection or call
> termination
> or transfer. The SIP session in question would be terminated.  From the
> SIP
> point of view, it seems to make sense to put the RTSP URL in a Contact:
or
> Also:, exact callflows and details to be worked out.
>
> b) Have a RTSP controlled stream as part of a conference ie an addon to
an
> existing or about-to-exist SIP session.
> There could be a number of interesting variants to this.
>
> One of them that follows the "SIP-UA knows RTSP" model:
> You can have one SIP UA be the RTSP client (ie the owner/controller of
the
> stream if you will), and direct the stream to each SIP UA in the
> session(could
> be via multicast as well). Other UAs dont need to know anything about
> RTSP. In
> which case SIP needs a way to inform UAs that  this new RTP stream will
be
> added
> to the session which I guess it should be able to do using re-INVITEs.
> If you go with the model that the RTSP server understands SIP, then RTSP
> provides a means to pass the conference ID to the RTSP server so that it
> can
> join the SIP session and link subsequent RTSP operations to the SIP
> session.
> Such IDs are opaque to RTSP protocol , so the usage would depend on the
> server
> and application in question.
>
> -Anup.
>
> > Linden deCarmo
> > Netspeak Corporation
> > 902 Clint Moore Road
> > Suite 104
> > Boca Raton, FL 33487
> >
> > > -----Original Message-----
> > > From: Kundan Singh [SMTP:kns10@cs.columbia.edu]
> > > Sent: Wednesday, March 15, 2000 3:10 PM
> > > To:   Linden deCarmo
> > > Cc:   'Dean Willis'; sip@lists.research.bell-labs.com
> > > Subject:      RE: SIP & RTSP interworking?
> > >
> > > --
> > > Kundan Singh     http://www.cs.columbia.edu/~kns10
> > >
> > > > This is an interesting suggestion, but the scenario I'm operating
> under
> > > is
> > > > this:
> > > >
> > > > UAC<->UAC (in this case the UAC really is an RTSP server), both
must
> > > > understand SIP and RTSP.  I need a way to tie the sessions between
> two
> > > > distinct protocols.  Is a viable alternative to embed the RTSP
SETUP
> > > message
> > > > as a SIP body for the INVITE?  This would clearly associate the
> > > sessions,
> > > > but I hate to have a proprietary solution that wouldn't work with
> other
> > > > products that understood SIP & RTSP.
> > >
> > > In that case, the UA may return the RTSP Contact: URI in (say) 3xx
> > > response.
> > >
> > >  UA1          UA2
> > >      INVITE
> > >  -------------->
> > >
> > >    3xx Moved
> > >    Contact:rtsp://server/resource
> > >  <-------------
> > >
> > >    SETUP rtsp://server/resource
> > >  -------------->
> > >  . . .
> > >
> > > Since UA here understand both SIP and RTSP it should not be
> > > a problem.
> > >
> > > If the SIP call is already established, and UA2 wants to
> > > switch to RTSP,  then (IMO) it may use blind call transfer
> > > (BYE,Also:rtsp://...) to ask UA1 to connect to RTSP
> > > server of UA2. Comments ??
> > >




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This is the second draft of the Call Flows document (first draft was titled
draft-johnston-sip-call-flows-00.txt before it was accepted as a working
group
draft).

The major changes from the first draft are:

     - Converted all figures to ASCII art.
     - Incorporated "SIP Telephony Service Examples with Call Flows"
(Sparks et al),
       draft-sparks-sip-service-examples-00, and also SIP Test Messages
from the 2nd
       Bakeoff (Rosenberg and Schulzrinne) into the document.
     - Fixed assorted typos.
     - Made references to ISUP explicitly ANSI and added some comments
       on ITU and ETSI.
     - Fixed CANCEL mistakes (use of tags, end to end acknowledgments,
etc.)
     - Added unique Authentication headers for each message.
     - Added Reliable Provisional Responses (PRACK) to scenario 4.1.2 for
       183 Session Progress
     - Added simple SIP to SIP call without proxies 3.1.1
     - Fixed "phone-context" and phone number handling in SIP URLs

Any comments on the draft are greatly appreciated.

Thanks to all the authors and the Call Flows Design Team.

Alan Johnston
MCI WorldCom

Internet-Drafts@ietf.org wrote:
>
> A New Internet-Draft is available from the on-line Internet-Drafts
directories.
> This draft is a work item of the Session Initiation Protocol Working
Group of the IETF.
>
>         Title           : SIP Telephony Call Flow Examples
>         Author(s)       : A. Johnston, S. Donovan, R. Sparks,C.
Cunningham,
>                           K. Summers,  D. Willis, J. Rosenberg,  H.
Schulzrinne,
>         Filename        : draft-ietf-sip-call-flows-00.txt
>         Pages           : 268
>         Date            : 06-Mar-00
>
> This document gives examples of SIP (Session Initiation Protocol)
> call flows for IP telephony and service examples.  Elements in these
> call flows include SIP User Agents and Clients, SIP Proxy and
> Redirect Servers, and Gateways to the PSTN (Public Switch Telephone
> Network).  IP telephony scenarios include SIP Registration, SIP to
> SIP calling, SIP to Gateway, Gateway to SIP, and Gateway to Gateway
> via SIP.  Call flow diagrams and message details are shown.  PSTN
> telephony protocols are illustrated using ISDN (Integrated Services
> Digital Network), ANSI ISUP (ISDN User Part), and FGB (Feature Group
> B) circuit associated signaling.  PSTN calls are illustrated using
> global telephone numbers from the PSTN and private extensions served
> on by a PBX (Private Branch Exchange).  Telephony service examples
> illustrated with call flows include call hold, transfer, forwarding,
> screening, and Find-Me.  Example SIP messages used for testing during
> SIP 'bakeoff' events include SIP 'torture test' messages, and
> messages with invalid parameters, methods, and tags.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-call-flows-00.txt
>
> Internet-Drafts are also available by anonymous FTP. Login with the
username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
>         "get draft-ietf-sip-call-flows-00.txt".
>
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
> Internet-Drafts can also be obtained by e-mail.
>
> Send a message to:
>         mailserv@ietf.org.
> In the body type:
>         "FILE /internet-drafts/draft-ietf-sip-call-flows-00.txt".
>
> NOTE:   The mail server at ietf.org can return the document in
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--
Kundan Singh     http://www.cs.columbia.edu/~kns10

> This is an interesting suggestion, but the scenario I'm operating under
is
> this:
>
> UAC<->UAC (in this case the UAC really is an RTSP server), both must
> understand SIP and RTSP.  I need a way to tie the sessions between two
> distinct protocols.  Is a viable alternative to embed the RTSP SETUP
message
> as a SIP body for the INVITE?  This would clearly associate the sessions,
> but I hate to have a proprietary solution that wouldn't work with other
> products that understood SIP & RTSP.

In that case, the UA may return the RTSP Contact: URI in (say) 3xx
response.

 UA1          UA2
     INVITE
 -------------->

   3xx Moved
   Contact:rtsp://server/resource
 <-------------

   SETUP rtsp://server/resource
 -------------->
 . . .

Since UA here understand both SIP and RTSP it should not be
a problem.

If the SIP call is already established, and UA2 wants to
switch to RTSP,  then (IMO) it may use blind call transfer
(BYE,Also:rtsp://...) to ask UA1 to connect to RTSP
server of UA2. Comments ??





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"Fairlie-Cuninghame, Robert" wrote:
>
> I have a couple of questions regarding the tag value added to the {To}
> header.
>
> The SIP RFC says that the tag value should be a UUID and (from a previous
> post) that the UUID should/may be constructed using the standard format.
The
> RFC also says that "The "tag" value MUST be globally unique and
> cryptographically random with at least 32 bits of randomness."
>
> So I take it that the tag can either be a UUID or 32bit random number ?

They are not mutually exclusive. I believe that the timestamp in the
UUID is sufficiently fine grained that a UUID has 32 bits of randomness.

>
> It has puzzled me why many traces have "tag=xxxxxxx" where xxxxxx is a
> decimal number usually much less than 2^31. Why decimal and so small ?
> [Whether the field is hex or decimal or UUID shouldn't matter since a
> case-insensitive string comparison of the field is used, correct?]

32 bit number is not the same thing at all as 32 bits of randomness. 32
bits of randomness means that if you  look at the tags as a random
process, the entropy of the probability mass function is 32 bits. A
simplistic way to view this is that the tags are random numbers between
1 and 2**32, with the probability of picking any particular one being 1
in 2**32. Thus, 1 is a perfectly valid tag and just as likely to show up
as 2**32 -1.

>
> What is the current best practice for adding a tag when there is only one
> Via line present? I think the spec uses an implicit MAY for this case but
> has this been upgraded/clarified to a SHOULD or MUST ? That is, a UAS
> SHOULD/MUST add a tag to final responses if there is one Via header (and
of
> course MUST if there are multiple Via's).

I think rfc2543bis will make tag addition a MUST. Having to special case
this is a big pain, as it turns out. We had it as a MAY in the first
place, since we wanted to keep the direct client to client case simple.

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

> > The current SIP BNF allows the general form of URI (or URI-reference?) to
> > be used in the request line and various headers. Observations and
> > questions:
> >
> > 1. A valid form of URI (RFC 2396)is the empty sequence since:
> >
> > URI-reference = [ absoluteURI | relativeURI ] [ "#" fragment ]
> >
> > This would permit the request line and various headers (To, From,
> > Contact. ..) without an address specification which doesn't seem to make
> > much sense. RFC 2396 specifies that the empty sequence URI refers to the
> > start of the current document; is there any corresponding abstraction for
> > SIP transcations?
>
> No. Please god no. Obviously RFC2396 specifies a much wider and more abstract form
> of URI than is commonly used in SIP. Not everything in 2396 is appropriate for
> every URI scheme. They are a set of guidelines that work more or less for the most
> common URI schemes in use today: http, ftp, telnet, mailto (with some quirks), etc.
>

Be that as it may, they're guidelines which are now an integral part of the SIP BNF.

>
> The Request-URI and Contact: URIs have different requirements for parsing than the
> To/From URIs. For example, a proxy can to a degree treat the To:/From: URIs
> as opaque strings but MUST understand the syntax/semantics of the Request-URI.
> This doesn't prevent you from doing rigorous parsing everywhere, but keep in
> mind that it doesn't necessarily need to be done everywhere.

Agreed, but at some point in a session the To:/From: URI's can quite possibly be put in a
Request_URI for a subsequent transaction that proxies will have to worry about.

>
>
> >
> > 2. The relative URI references a resource relative to a base URI of a
> > hierarchial scheme. So a similar question: is this a useful concept for
> > SIP sessions?
> >
>
> The only contexts where I would think this might be useful would be in
> 1) The headers parameter of a SIP URI (for the From: and possibly To: headers)
> 2) A SIP "cookie" similar to an HTTP-cookie. This hasn't really been investigated
>    much but would be interesting.
>
> In other words, I don't think it is generally useful in the usual places in a SIP
> message (To/From/Contact/Request-URI). But it is potentially useful at the user interface
> level or as part of some service logic.

As part of a SIP message? I guess I can think of many examples where you want to map user
friendly "addresses" to URI's in SIP messages, but that's out of scope for this discussion.

>
>
> > 3. Finally RFC 2396 states (my emphasis):
> >   "When a URI reference is used to perform a retrieval action on the
> >    identified resource, the optional fragment identifier, separated from
> >    the URI by a crosshatch ("#") character, consists of additional
> >    reference information to be interpreted by the user agent after the
> >    retrieval action has been successfully completed.  As such, it is not
> >    part of a URI, but is often used in conjunction with a URI."
> >
>
> True, but this treated in very different ways by different web browsers. Some
> treat it as part of the URI, others use it for "post-processing".
> It's not really used in a standard way for SIP, but as stated above it really is up
> to the UserAgent's interpretation. THIS IS A GOOD THING.

I'm not arguing the merits of the URI syntax, just how it's used in SIP messages.

>
>
> >
> > So this very general URI seems to have a number of properties which don't
> > seem to be relevant when used in place of a SIP URL. Since generality
> > often has a price either in implementation or specification, i.e., lots
> > of semantic constraints, does it make sense to restrict the syntactic
> > form of URI used in place of a SIP URL, e.g., to the absolute-URI form?
> >
>
> Well, if you really want to dig into this, the SIP URI as specified in 2543 is
> actually a much different beast. Technically the entire SIP URL according to 2396
> falls under the "opaque_part" category of absolute URIs. What does this mean? It means
> that the syntax AND semantics of a SIP URI are NOT defined in 2396 at all but left
> to the definition in 2543. So from this perspective, much of 2396 does not apply to sip:
> URIs. I don't know if this was the intention of the SIP authors or not. From a common
> sense perspective, however, a SIP URI has all the properties of the URLs as defined
> in 2396: it is a transcribable uniform resouce identifier :)
>
> I see no benefit in restricting the syntax of a SIP URI. If you "ignore" the fact that
> the SIP URI should be treated as opaque, you can parse the SIP URI just like an
> http, ftp, telnet, etc. URI. In this sense, it would actually place more of a burden
> on an implementation to prescribe special syntax restrictions on a SIP URI. Furthermore,
> I'd prefer to leave open the possibility for future SIP extensions which might need
> something like a relative SIP URI.
>

Since the URI's we're discussing can end up as Request-URI's, I think the current syntax is
just too general. At the very least, insisting that the scheme be present doesn't seem
unreasonable. And I guess I can't quite grasp the notion of a relative URL in this context,
so maybe a clarifying example would help.

>
> >
> > Rick Workman
> > Nortel Networks
> > Ottawa
>
> --
> Sean Olson <sean.olson@ericsson.com>
> Ericsson Inc.
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--------------880028A09547835F614D226E
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<tt></tt>&nbsp;<tt></tt>
<p><tt>Sean Olson wrote:</tt>
<blockquote TYPE=CITE><tt>> The current SIP BNF allows the general form
of URI (or URI-reference?) to</tt>
<br><tt>> be used in the request line and various headers. Observations
and</tt>
<br><tt>> questions:</tt>
<br><tt>></tt>
<br><tt>> 1. A valid form of URI (RFC 2396)is the empty sequence since:</tt>
<br><tt>></tt>
<br><tt>> URI-reference = [ absoluteURI | relativeURI ] [ "#" fragment
]</tt>
<br><tt>></tt>
<br><tt>> This would permit the request line and various headers (To, From,</tt>
<br><tt>> Contact. ..) without an address specification which doesn't seem
to make</tt>
<br><tt>> much sense. RFC 2396 specifies that the empty sequence URI refers
to the</tt>
<br><tt>> start of the current document; is there any corresponding abstraction
for</tt>
<br><tt>> SIP transcations?</tt><tt></tt>
<p><tt>No. Please god no. Obviously RFC2396 specifies a much wider and
more abstract form</tt>
<br><tt>of URI than is commonly used in SIP. Not everything in 2396 is
appropriate for</tt>
<br><tt>every URI scheme. They are a set of guidelines that work more or
less for the most</tt>
<br><tt>common URI schemes in use today: http, ftp, telnet, mailto (with
some quirks), etc.</tt>
<br><tt></tt>&nbsp;</blockquote>
<tt>Be that as it may, they're guidelines which are now an integral part
of the SIP BNF.</tt>
<blockquote TYPE=CITE><tt></tt>&nbsp;
<br><tt>The Request-URI and Contact: URIs have different requirements for
parsing than the</tt>
<br><tt>To/From URIs. For example, a proxy can to a degree treat the To:/From:
URIs</tt>
<br><tt>as opaque strings but MUST understand the syntax/semantics of the
Request-URI.</tt>
<br><tt>This doesn't prevent you from doing rigorous parsing everywhere,
but keep in</tt>
<br><tt>mind that it doesn't necessarily need to be done everywhere.</tt></blockquote>
<tt>Agreed, but at some point in a session the To:/From: URI's can quite
possibly be put in a Request_URI for a subsequent transaction that proxies
will have to worry about.</tt>
<blockquote TYPE=CITE><tt></tt>&nbsp;<tt></tt>
<p><tt>></tt>
<br><tt>> 2. The relative URI references a resource relative to a base
URI of a</tt>
<br><tt>> hierarchial scheme. So a similar question: is this a useful concept
for</tt>
<br><tt>> SIP sessions?</tt>
<br><tt>></tt><tt></tt>
<p><tt>The only contexts where I would think this might be useful would
be in</tt>
<br><tt>1) The headers parameter of a SIP URI (for the From: and possibly
To: headers)</tt>
<br><tt>2) A SIP "cookie" similar to an HTTP-cookie. This hasn't really
been investigated</tt>
<br><tt>&nbsp;&nbsp; much but would be interesting.</tt><tt></tt>
<p><tt>In other words, I don't think it is generally useful in the usual
places in a SIP</tt>
<br><tt>message (To/From/Contact/Request-URI). But it is potentially useful
at the user interface</tt>
<br><tt>level or as part of some service logic.</tt></blockquote>
<tt>As part of a SIP message? I guess I can think of many examples where
you want to map user friendly "addresses" to URI's in SIP messages, but
that's out of scope for this discussion.</tt>
<blockquote TYPE=CITE><tt></tt>&nbsp;<tt></tt>
<p><tt>> 3. Finally RFC 2396 states (my emphasis):</tt>
<br><tt>>&nbsp;&nbsp; "When a URI reference is used to perform a retrieval
action on the</tt>
<br><tt>>&nbsp;&nbsp;&nbsp; identified resource, the optional fragment
identifier, separated from</tt>
<br><tt>>&nbsp;&nbsp;&nbsp; the URI by a crosshatch ("#") character, consists
of additional</tt>
<br><tt>>&nbsp;&nbsp;&nbsp; reference information to be interpreted by
the user agent after the</tt>
<br><tt>>&nbsp;&nbsp;&nbsp; retrieval action has been successfully completed.&nbsp;
As such, it is not</tt>
<br><tt>>&nbsp;&nbsp;&nbsp; part of a URI, but is often used in conjunction
with a URI."</tt>
<br><tt>></tt><tt></tt>
<p><tt>True, but this treated in very different ways by different web browsers.
Some</tt>
<br><tt>treat it as part of the URI, others use it for "post-processing".</tt>
<br><tt>It's not really used in a standard way for SIP, but as stated above
it really is up</tt>
<br><tt>to the UserAgent's interpretation. THIS IS A GOOD THING.</tt></blockquote>
<tt>I'm not arguing the merits of the URI syntax, just how it's used in
SIP messages.</tt>
<blockquote TYPE=CITE><tt></tt>&nbsp;<tt></tt>
<p><tt>></tt>
<br><tt>> So this very general URI seems to have a number of properties
which don't</tt>
<br><tt>> seem to be relevant when used in place of a SIP URL. Since generality</tt>
<br><tt>> often has a price either in implementation or specification,
i.e., lots</tt>
<br><tt>> of semantic constraints, does it make sense to restrict the syntactic</tt>
<br><tt>> form of URI used in place of a SIP URL, e.g., to the absolute-URI
form?</tt>
<br><tt>></tt><tt></tt>
<p><tt>Well, if you really want to dig into this, the SIP URI as specified
in 2543 is</tt>
<br><tt>actually a much different beast. Technically the entire SIP URL
according to 2396</tt>
<br><tt>falls under the "opaque_part" category of absolute URIs. What does
this mean? It means</tt>
<br><tt>that the syntax AND semantics of a SIP URI are NOT defined in 2396
at all but left</tt>
<br><tt>to the definition in 2543. So from this perspective, much of 2396
does not apply to sip:</tt>
<br><tt>URIs. I don't know if this was the intention of the SIP authors
or not. From a common</tt>
<br><tt>sense perspective, however, a SIP URI has all the properties of
the URLs as defined</tt>
<br><tt>in 2396: it is a transcribable uniform resouce identifier :)</tt><tt></tt>
<p><tt>I see no benefit in restricting the syntax of a SIP URI. If you
"ignore" the fact that</tt>
<br><tt>the SIP URI should be treated as opaque, you can parse the SIP
URI just like an</tt>
<br><tt>http, ftp, telnet, etc. URI. In this sense, it would actually place
more of a burden</tt>
<br><tt>on an implementation to prescribe special syntax restrictions on
a SIP URI. Furthermore,</tt>
<br><tt>I'd prefer to leave open the possibility for future SIP extensions
which might need</tt>
<br><tt>something like a relative SIP URI.</tt>
<br><tt></tt>&nbsp;</blockquote>
<tt>Since the URI's we're discussing can end up as Request-URI's, I think
the current syntax is just too general. At the very least, insisting that
the scheme be present doesn't seem unreasonable. And I guess I can't quite
grasp the notion of a relative URL in this context, so maybe a clarifying
example would help.</tt>
<blockquote TYPE=CITE><tt></tt>&nbsp;
<br><tt>></tt>
<br><tt>> Rick Workman</tt>
<br><tt>> Nortel Networks</tt>
<br><tt>> Ottawa</tt><tt></tt>
<p><tt>--</tt>
<br><tt>Sean Olson &lt;sean.olson@ericsson.com></tt>
<br><tt>Ericsson Inc.</tt><tt></tt>
<p><tt>_______________________________________________</tt>
<br><tt>SIP mailing list</tt>
<br><tt>SIP@lists.bell-labs.com</tt>
<br><tt><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></tt></blockquote>
<tt></tt></html>

--------------880028A09547835F614D226E--



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From sip-admin@lists.bell-labs.com  Mon May  8 23:13:41 2000
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A few low-level comments:

- I'm not sure "TRANSFER" is the right verb here, since transfer implies
moving something from one place to another. All TRANSFER does is
effectively ask the recipient to send another INVITE. It would be nice
if the same action could be used (say) for mesh conferences. While I
agree that using methods is better than adding headers, we don't want
one method per feature, if we can help it. (I.e., we should abstract
what's happening at the signaling layer and then decide what operations
are needed for that rather than just map feature to request name.)

Maybe PLEASE-INVITE or something along these lines. That way, if we need
a 'remote BYE', we can add 'PLEASE-BYE'.

- Using TRANSFER instead of INVITE with Also does have one major
disadvantage, since the originator of the request will retransmit with
T2 (4 seconds).

- According to the rules for feature names, IETF-based extensions are
single words, to avoid any possible confusion with proprietary
extensions having inverted domain names. Thus, it should probably just
be Requires: transfer rather than cc.transfer. (On the other hand, it's
not clear this is needed at all - we don't need Require for new methods,
since they already have a mechanism to say "sorry, je ne comprend pas").

- Telepathy transfer:

"UA receiving a well-formed TRANSFER request SHOULD request approval
   from the user to proceed. In the absence of that request, or upon
                                    ^^^^^^^^
--> If there's no request, the UAS spontaneously emits an INVITE?

   receiving approval from the user, the UA MUST submit an INVITE to
   the resource identified by the Transfer-To: header using the Call-ID
   from the TRANSFER request."

Should probably be UAS, to make things clearer.




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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        John Mainwaring <crm312a@nortelnetworks.com>
Cc: "Mark D. Foster" <mark.foster@npac.com>,
        iptel@lists.research.bell-labs.com,
        "'James Yu'" <james.yu@neustar.com>, sip@lists.research.bell-labs.com
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Date: Mon, 8 May 2000 14:10:12 +0530
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Subject: [SIP] URL Generalization for Portability (was TRIP)
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Jonathan Rosenburg wrote:
> I wonder if we can't think about this problem in more general terms.

yes, please!


> In the SIP model, each proxy owns some piece of the namespace
> (engineering.dynamicsoft.com, for example), and is therefore responsible
> for the definition of the user space (jdrosen@) within that domain.
> Thats why each proxy can, an usually will, perform some kind of URI
> translation independent from other proxies. Our underlying assumption
> was that usernames only have meaning within their domain. For example,
> with sip:jdrosen@dynamicsoft.com, only a dynamicsoft.com proxy knows how
> to handle jdrosen. There might be jdrosen's in other domains that aren't
> me. However, in the case of telephone numbers, the proxy that owns the
> domain does not necessarily own the user space. sip:5551212@gateway.com,
> for example, uses a user name that is not "owned" by gateway.com, in the
> sense that this name has valid semantics outside of gateway.com. Other
> domains, like wcom.com, would know how to handle this number as well.
> Its possible that we might even see things besides phone numbers as user
> names that are valid across multiple domains. An example is (heaven
> forbid), social security numbers (in the US, at least):

The underlying problem here is differentiating between a phone-number and a
service-name-which-looks-like-a-number. Yes, the user=phone gives a hint on
nature od address, the "+" encodding gives a hint on the scope of address,
but it is still muddy.

There was some discussion of this problem wrt DCS during the November
interim meeting. It was suggested that a cross-domain destination which is
a
phone number could be better represented by a tel: URL than by confusing
the
scope of the user part of a sip: URL. This simplilifies life greatly.

Something like

     INVITE premiumservice@gateway.carrier.com
     To: tel:5551212;trans=lnp?gdbs

could be used to indicate to gateway.carrier.com that premium gateway
service is requested towards the PSTN location identified by telephone
number 5551212, that LNP and GDBS translation have been applied, and that
the phone number is of local scope for the gateway. Incidentally, I think I
just noticed a minor anomaly in the tel-url grammar -- don't see how to
encode more than one future-extension field . . .

--
Dean


---------
This message came from the IETF IPTEL Working Group Mailing List.




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To: Anoop Tripathi <Anoop_Tripathi@mw.3com.com>
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Anoop Tripathi wrote:
>
> Hi,
>
> Is there a draft or a generally accepted fault tolerance scheme for SIP
Proxy
> Servers?

There is no draft. I'm not even sure there should be (at least not
standards track). How a server handles fault tolerance is an
implementation issue. THere are numerous capabilities of SIP you can use
to build robust systems. We have discussed several on the list,
including multicast, statelessness, usage of domain names instead of IP
addresses, and so on.

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




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Dean,

This is an interesting suggestion, but the scenario I'm operating under is
this:

UAC<->UAC (in this case the UAC really is an RTSP server), both must
understand SIP and RTSP.  I need a way to tie the sessions between two
distinct protocols.  Is a viable alternative to embed the RTSP SETUP
message
as a SIP body for the INVITE?  This would clearly associate the sessions,
but I hate to have a proprietary solution that wouldn't work with other
products that understood SIP & RTSP.

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

> -----Original Message-----
> From:   Dean Willis [SMTP:dean.willis@wcom.com]
> Sent:   Wednesday, March 15, 2000 2:10 PM
> To:     Linden deCarmo; sip@lists.research.bell-labs.com
> Subject:     RE: SIP & RTSP interworking?
>
>
> Here's what I would suggest:
>
> Build a SIP UAS that understands the RTSP end and contains stream
> management
> logic. This UAS either reinvites to establish the link to RTSP streams,
or
> copies the RTP from each stream back into the media path for the remote
> UA.
>
> I know of some people(s) who are already using each approach to build a
> voice mail system that can use a standard RTSP server as a media store.
> Makes unified messaging with URL passing MUCH easier.
>
> --
> dean
>
> > -----Original Message-----
> > From: owner-sip@lists.research.bell-labs.com
> > [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Linden
> > deCarmo
> > Sent: Tuesday, March 14, 2000 7:27 AM
> > To: 'sip@lists.research.bell-labs.com'
> > Subject: SIP & RTSP interworking?
> >
> >
> > What would be the proper (or suggested) procedure to tie a SIP
> > session to an
> > RTSP session?
> >
> > What I'd like to do is setup a session with SIP (via an INVITE), then
> > control streaming aspects via RTSP (i.e. play, pause, stop etc.).
> >  However,
> > its not clear to me how I tie the RTSP session id to the previously
> > established SIP session.
> >
> > Thanks.
> >
> > Linden deCarmo
> > Netspeak Corporation
> > 902 Clint Moore Road
> > Suite 104
> > Boca Raton, FL 33487
> >
> >




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Also, please note that an IPR statement has been made regarding this
draft. It can be found at:
http://www.ietf.org/ietf/IPR/DIALOGIC-SIP-DMCS

-Jonathan R.

Jeff.Mark@Dialogic.com wrote:
>
> Hi all,
>
> Just wanted to let you know of a new draft.  Below is the announcement in
> case you missed it.  We know there is other work going on in the call
> control services area (draft-ietf-sip-cc-01, and new drafts
> draft-campbell-sip-cc-framework-00 and draft-sparks-sip-cc-transfer-00)
> and hope the ideas presented in our draft below are of interest to
others.
>
> Comments are appreciated.
>
> Thanks!
>
> -Jeff
>

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




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Subject: [SIP] Re: uploading services from the user's  home network to a sip
 server of  the visited network within a 3G.IP environement
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"Liess, Laura" wrote:
>
> Hi,
>
> after doing a very good job by deciding to take SIP for call control for
multimedia services within 3G.IP >networks, the 3G.IP operators have a lot
of open items concerning the architecture, one of them being which >network
controls the services for a roaming user: the home network, the visited
network or shared control. >The concrete question: how can I upload at the
begin or during the call the roamimg user's services >available in the home
network into the SIP proxy of the visited network? As far as I know, the
CPL and >REGISTER helps me to upload services only from the UA to the
proxy.

I'm not an expert in wireless, so I'm not certain of the scope of the
services we are talking about. Mid call services, for example, are not
currently within the scope of CPL. However, there is nothing which
prevents the CPL-upload-in-register mechanism from being used by a
server to upload scripts to another server. Registrations, in general,
can be used for third party controls. The From field identifies the
party initiating the registration.

-Jonathan R.

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

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From sip-admin@lists.bell-labs.com  Mon May  8 23:29:40 2000
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To: James Yu <james.yu@neustar.com>
Cc: John Mainwaring <crm312a@nortelnetworks.com>,
        Mark Foster <mark.foster@neustar.com>,
        iptel@lists.research.bell-labs.com, sip@lists.research.bell-labs.com
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James Yu wrote:
>
> For overlap signaling, a GW or LS that receives a call with partial
address
> information may not know that the address info. is complete.  If it
launches
> a query (DNS or NPDB), the entity that has the "number portability" info.
> won't be able to find the info. because the address info. is incomplete.
It
> seems that either this LS or GW may wait a little while to see whether
there
> are subsequent address info. before launching the query, or it can launch
a
> query (DNS or NPDB-like), receives an error (hope that it is specific
such
> as incomplete address instead of something ambiguous), and then wait for
> additional address info. before timing out.

I think Dean had sent out some pseudo-code some time back on how trip
and enum and overlap interact, and this was more or less what the code
suggested. I believe he thought it was a bad thing, and so do I.

Overlap dialing seems fundamentally in conflict with LNP or enum queries
at the local outbound proxy.

To do LNP, the local outbound proxy needs to have the entire string of
digits before it can perform the query. But, overlap seems to imply that
it won't ever know the entire string of digits, as it doesn't have
knowledge of those numbering plans. It just forwards the request to some
other proxy or gateway that does.

Sooo, it would seem that a proxy should only use enum or LNP for numbers
it knows to be complete. This means that for overlap dialing, the LNP
query will happen close to the destination (possibly even in the PSTN),
not unlike the N-1 case we've been discussing. However, for en-bloc, it
can happen anywhere. This does seem to imply that a proxy needs to know
whether the number in the request URI of an incoming request is en-bloc
or overlap. It could make this determination if it knew the structure of
the numbering plans, but thats unlikely. Might we need a parameter
somewhere (ugh)? I really hate to crowd SIP with things to support old
legacy technologies like overlap, so if we can avoid this, I'd be very
happy.



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

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Subject: [SIP] New I-D: SIP Security using CHAP-Password
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Hi guys,

Attached is an initial I-D on SIP Security using CHAP-Password.  This
I-D allows authentication of SIP clients using a backend Radius (RFC
2138) server.

I'd appreciate comments/criticisms/suggestions and I'd like to trigger a
discussion on the outstanding issues.

Enjoy!

Bryan

Bryan J. Byerly
byerly@cisco.com
(919) 392-7091



Internet Engineering Task Force                         Bryan J. Byerly
Internet Draft                                           David Williams
draft-byerly-sip-radius-00.txt                            Cisco Systems
March, 2000
Expires: September, 2000





                 SIP Security using CHAP-Password

Status of this Memo

  This document is an Internet-Draft and is in full conformance with
  all provisions of Section 10 of RFC2026.

  Internet-Drafts are working documents of the Internet Engineering
  Task Force (IETF), its areas, and its working groups. Note that
  other groups may also distribute working documents as Internet-
  Drafts.

  Internet-Drafts are draft documents valid for a maximum of six
  months and may be updated, replaced, or obsoleted by other documents
  at any time. It is inappropriate to use Internet-Drafts as
  reference material or to cite them other than as "work in progress."

  The list of current Internet-Drafts can be accessed at
  http://www.ietf.org/ietf/lid-abstracts.txt.

  The list of Internet-Draft Shadow Directories can be accessed at
  http://www.ietf.org/shadow.html.

Abstract

  This document describes a proposed extension to SIP.  This
  document proposes using an alternative SIP security mechanism
  for use in Proxy-Authorization or Authorization headers in order
  to support SIP client Authentication using backend RADIUS servers.

  The introduction of this extension would allow a SIP proxy
  (or called SIP client) to authenticate a SIP client using a backend
  RADIUS server.










Byerly/Williams      draft-byerly-sip-radius-00.txt               Page 1

Internet Draft       SIP Security using CHAP-Password         March 2000

1 Introduction

  Some ISPs currently use RADIUS servers to authenticate
  (and implicitly authorize) dialup users for PPP service.
  It would be advantageous to allow the re-use of this same
  RADIUS infrastructure for SIP client authentication.

  Although the currently defined mechanisms for SIP client
  authentication [section 3.2.2.2 of RFC 2617] and RADIUS
  authentication using a User-Password or CHAP-Password
  [sections 5.2, 5.3 of RFC 2138] both use MD5, they run MD5
  across differently formatted messages.  There are two
  approaches to solving this problem.  One is to extend
  RADIUS to support HTTP-Digest; the other is to extend
  the SIP list of authentication schemes to support a CHAP-Password.
  This document proposes extending the SIP list of authentication
  schemes to support a CHAP-Password.


2 Definitions

  The definitions of several terms used in this document follow:

  nonce

      A nonce is a octet string that is uniquely generated each time a
      request is made.  It is recommended that a nonce be constructed
      to exhibit global and temporal uniqueness.

      The SIP specification [SIP] calls this a "nonce-value"
      (Section 3.2.1 of [DIG]).

      The RADIUS specification calls this a (random) challenge.
      (Section 2.2 of [RAD])
      In RADIUS, the nonce can be placed in the Request Authenticator
      (Section 4.2 of [RADIUS]) or in the CHAP-Challenge attribute.
      (Section 5.40 of [RADIUS])

      The CHAP Response in the CHAP-Password and the
      nonce-value in the HTTP-Digest use a 16-octet nonce.


  sequence number

     A sequence number is a monotonically increasing integer.
     Sequence numbers allow detection of replays.

     The "nonce-count" of the HTTP-Digest is a 32-bit sequence number
     (formatted as 8 hex digit characters)

     The "Chap-ID" in the CHAP-Password is a 1 octet sequence number.

Byerly/Williams      draft-byerly-sip-radius-00.txt               Page 2

Internet Draft       SIP Security using CHAP-Password         March 2000

  shared secret

      A secret shared between two entities.

      In this document, we assume that:

      - A user shares a secret (i.e. password) with
        the RADIUS server.

      - The PPP NAS (or SIP proxy) also shares a secret
        with the RADIUS server.


3 Analogous Model - PPP

  When a SIP proxy is used for user authentication
  an analogy can be drawn from PPP message flows.

3.1 Message flow for PPP user authentication with Radius backend:

  dialup                PPP                   RADIUS
  user                  NAS                   server
  |                      |                       |
  |--modem connection--->|                       |
  |                      |                       |
  |  PPP                 |                       |
  |<-Configure-Request---|                       |
  |                      |                       |
  |  PPP                 |                       |
  |--Configure-Ack------>|                       |
  |                      |                       |
  |                      |--Access-Request------>|
  |                      |                       |
  |                      |<-Access-Challenge-----|
  |                      |                       |
  |<-CHAP Challenge------|                       |
  |                      |                       |
  |--CHAP Response------>|                       |
  |                      |                       |
  |                      |--Access-Request------>|
  |                      |                       |
  |                      |<-Access-Accept--------|
  |                      |                       |
  |                      |                       |








Byerly/Williams      draft-byerly-sip-radius-00.txt               Page 3

Internet Draft       SIP Security using CHAP-Password         March 2000

3.2 Message flow for SIP user authentication with Radius backend:

  SIP                 SIP                     RADIUS
  client              proxy                   server
  |                      |                       |
  |--[1] INVITE--------->|                       |
  |                      |                       |
  |                      |--[2] Access-Request-->|
  |                      |                       |
  |                      |<-[3] Access-Challenge-|
  |                      |                       |
  |<-[4] 407-------------|                       |
  | (Proxy-Authenticate:)                        |
  |                      |                       |
  |--[5] INVITE--------->|                       |
  |(Proxy-Authorization:)                        |
  |                      |                       |
  |                      |--[6] Access-Request-->|
  |                      |                       |
  |                      |<-[7] Access-Accept----|
  |                      |                       |
  |                      |--[8] INVITE------------------>
  |                      |                       |
  .
  .
  .
  |<--200---------------|<--200------------------------

  When a PPP client authentication failure occurs, in some
  cases the PPP NAS implementation terminates the link.
  However, other PPP NAS implementations may choose to allow
  the client to continue, but with a filtered list of services.
  A PPP NAS may allow traffic which lets the user update his credentials
  (such as email to the sysadmin).

  Similarly, a SIP proxy server may wish to allow the user to place
  calls to the ISP's home office (to obtain updated credentials).
  A SIP proxy server may also wish to allow 911 calls to complete.


4 Current PAP/CHAP/SIP/HTTP Authentication mechanisms

  The following sections briefly describe the current mechanisms
  used for user authentication in PPP (PAP/CHAP) and HTTP/SIP
  (Basic and Digest).







Byerly/Williams      draft-byerly-sip-radius-00.txt               Page 4

Internet Draft       SIP Security using CHAP-Password         March 2000

4.1 PAP authentication mechanism

    Why not use RADIUS User-Password?

    The RADIUS User-Password attribute is calculated as:

        User-Password = md5hash(NAS-secret, nonce) XOR user-password

    If a PPP user sends his password in cleartext (eg. using PAP), then
    the PPP server can calculate the User-Password attribute of the
    Access-Request to authenticate the user.

    It is undesirable for a SIP user to send his password in cleartext.

    If the user does NOT send his password in cleartext,
    the User-Password cannot be calculated by either the PPP client
    (because he doesn't know the NAS-secret) or the NAS
    (because he doesn't know the user-password).

4.2 CHAP authentication mechanism

    PPP defines usage of the CHAP-Password (as an alternative
    to User-Password) to avoid cleartext transmission of the
    users's password.

    When a CHAP-Password is used a cleartext sequence number, cleartext
    nonce, and the following MD5 hash are transmitted by the client:

        md5hash(seqnum, user-password, nonce)

    The nonce can be generated by the client or the server.
    If the nonce is generated by the client, the server may choose
    to accept it or may challenge the client with a new nonce.

4.3 SIP/HTTP authentication mechanisms:

    SIP and HTTP define two basic authentication mechanisms.
    HTTP-Basic and HTTP-Digest.  Usage of HTTP-Basic involves
    sending the the user's password in cleartext, and is thus
    undesirable.

    The currently defined mechanisms for SIP client authentication
    using HTTP-Digest are taken from section 3.2.2.2 of RFC 2617
    and the hash constructions are repeated here for
    clarity:

    If the directive's value is "MD5" or is unspecified, then A1 is:

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

    If the directive's value is "MD5-sess", then A1 is

Byerly/Williams      draft-byerly-sip-radius-00.txt               Page 5

Internet Draft       SIP Security using CHAP-Password         March 2000

    calculated only once - on the first request by the client following
    receipt of a WWW-Authenticate challenge from the server.  It uses
    the server nonce from the challenge, and the first client nonce
    value to construct A1 as follows:

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

    This creates a 'session key' for the authentication of subsequent
    requests and responses which is different for each "authentication
    session", thus limiting the amount of material hashed with any one
    key. [RFC 2617]

5 Interaction/Mapping problems

  There are two problems:

  5.1 CHAP-Password construction problem:

      If a SIP proxy receives an HTTP-Digest from a SIP client
      (without CHAP-Password support), the SIP proxy is unable
      to construct a CHAP-Password.  This is because the SIP proxy
      doesn't have access to the client's password.
      The SIP proxy only has access to a hash of the client's password,
      and (as dicussed above) this hash is computed across a
      message whose format is different than the RADIUS server expects.

      Nor can the SIP proxy simply forward the hash calculated in
      the HTTP-Digest:

  5.2 Message mapping problem:

      Since the message format over which a hash is computed
      is different for the CHAP-Password than the message format
      used for the HTTP-Digest "MD5" or "MD5-sess" algorithms,
      a RADIUS server could not verify a proxied HTTP-Digest
      (which uses either the "MD5" or "MD5-sess" algorithms).
      The RADIUS server would discard such a HTTP-Digest
      formulated hash as invalid.

  Here is the proposed solution to these problems:

6 SIP Security using CHAP-Password

  To solve these problems, we specify an additional mechanism
  for SIP security which uses a CHAP-Password.  CHAP-Password
  can either be used for endpoint-to-endpoint authentication
  (when used in WWW-Authenticate and Authorization) or for
  endpoint-to-proxy authentication (when used in Proxy-Authenticate
  and Proxy-Authorization).

Byerly/Williams      draft-byerly-sip-radius-00.txt               Page 6

Internet Draft       SIP Security using CHAP-Password         March 2000

6.1 The WWW-Authenticate Response Header

    When a CHAP-Password is used for SIP security,
    the WWW-Authenticate Response Header (3.2.2 of RFC 2617)
    is defined as:

WWW-Authenticate = "WWW-Authenticate" ":" "CHAP-Password" chap-challenge
chap-challenge   = * (";" chap-params )
chap-params      = chap-username | chap-algorithm | chap-id | nonce
chap-algorithm   = "algorithm" "=" ( "MD5" | token )
chap-username    = quoted-string
chap-id          = "id" "=" + ( digit )
chap-nonce       = "nonce" "=" nonce-value
chap-nonce-value = <"> 32LHEX <">
LHEX             = "0" | "1" | "2" | "3" |
                   "4" | "5" | "6" | "7" |
                   "8" | "9" | "a" | "b" |
                   "c" | "d" | "e" | "f"

    chap-algorithm: A string indicating the authentication
                    method to be used.

    chap-username:  A string containing the user name.

    chap-id:        The CHAP Identifier is a one octet sequence number.

    nonce:          A string of 32 hex digits.  The contents of the
                    nonce are implementation dependent.  The quality
                    of the implementation depends on a good choice.
    Example:

    WWW-Authenticate: CHAP-Password ;username="byerly" ;algorithm="MD5"
      ;id=0 ;nonce="10131973aaa511bb05261975aaa505fb"

    The chap-username is copied from the User-Name attribute
    of the Access-Challenge message received from the RADIUS server.

    The chap-id is copied from the (1-octet) Identifier field of the
    Access-Challenge message received from the RADIUS server.

    The chap-nonce-value is copied from the Access-Challenge message
    from the RADIUS server (from the CHAP-Challenge attribute if
    present, otherwise from the Request Authenticator).









Byerly/Williams      draft-byerly-sip-radius-00.txt               Page 7

Internet Draft       SIP Security using CHAP-Password         March 2000

6.2 The Authorization Request Header

    When challenged, the SIP client is expected to retry the request,
    passing an Authorization header line, which is defined as follows:

Authorization = "Authorization" ":" "CHAP-Password" chap-response-line
chap-response-line   =  * (";" chap-response-params )
chap-response-params = chap-username | chap-id | nonce | chap-response
chap-response        = "response" "=" chap-response-value
chap-response-value  = <"> 32LHEX <">

    chap-response-value: A string of 32 hex digits computed as defined
    in Section 4.1 of RFC1994, which proves that the user knows a
    password.

    Example:

    Authorization: CHAP-Password ;username="byerly" ;id=0
      ;nonce="10131973aaa511bb05261975aaa505fb"
      ;response="f53a66e43c12a383aa65219ec873ce35"

    The client MUST increment the CSeq header before resubmitting
    the request.

    A server MAY be configured not to generate nonces only if replay
    attacks are not a concern.

    The Response Value (chap-response-value) of the CHAP-Password is
    computed per Section 4.1 of RFC 1994 [CHAP].  The 16-octet
    Response Value of the CHAP-Password should be formatted as
    32 hex digits and placed in the "chap-response-value" of
    the Authorization request.

    The chap-id should be placed in the (1-octet) Identifier field
    of the Access-Request message to the RADIUS server.  The chap-id
    should also be placed in the (1-octet) CHAP Identifier field
    of the CHAP-Password attribute of the Access-Request message
    to the RADIUS server.  (See sections 3, 5.3, [RAD])

    The nonce-value SHOULD be placed in the Request Authenticator
    of the Access-Request message to the RADIUS server.
    (See section 3, [RAD])
    Alternatively, the nonce-value MAY be placed in a CHAP-Challenge
    attribute in the Access-Request message to the RADIUS server.
    (See section 5.3, [RAD])

    The chap-response-value should be placed in the 16-octet
    String field of the CHAP-Password attribute in the Access-Request
    message to the RADIUS server.  (See section 5.3, [RAD])




Byerly/Williams      draft-byerly-sip-radius-00.txt               Page 8

Internet Draft       SIP Security using CHAP-Password         March 2000

7 Proxy-Authenticate and Proxy-Authorization

  The CHAP-Password authentication scheme may also be used
  for authenticating users to proxies.

8 Security Considerations

  Security issues are the primary topic of this RFC.

  The security issues for this document are the same as those
  in the Security Considerations sections of RFC 1994 [CHAP]
  and RFC 2617 [DIG].

9 Further Examples

  Only the relevant headers have been included in the following
  examples.

9.1 User Authentication using backend RADIUS server -
    With Server Challenge.

     [1] SIP Client to SIP proxy server:

          INVITE sip:+19195551212@cisco.com SIP/2.0
          From: sip:+19195551234@domain.com
          To: sip:+19195551212@cisco.com
          Call-ID: 12345600@cisco.com
          CSeq: 1 INVITE
          Proxy-Authorization: CHAP-Password
            ;username="byerly"
            ;algorithm="MD5"
            ;id=0
            ;nonce="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa"
            ;response="bbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbb"
          Content-Type: application/sdp

NOTE: The Proxy-Authorization header sent in the first message
      may have been cached from a previous exchange with the
      SIP proxy.  If the SIP client does not place a
      Proxy-Authorization header in the INVITE, the RADIUS server will
      (transitting through SIP proxy) challenge him with a
      new nonce.

     [2] SIP proxy server to RADIUS server:

          Code = 1        (Access-Request)
          ID = 0
          Length = 71
          Request Authenticator = {16 octet random number also used as
                                   CHAP challenge
                                   (aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa)}

Byerly/Williams      draft-byerly-sip-radius-00.txt               Page 9

Internet Draft       SIP Security using CHAP-Password         March 2000

          Attributes:
              User-Name = "byerly"
              CHAP-Password = {1 octet CHAP ID (00) followed by 16 octet
                               CHAP response
                               (bbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbb)}
              NAS-IP-Address = 192.168.1.16
              NAS-Port = 5

     [3] RADIUS server to SIP proxy server:

          Code = 11       (Access-Challenge}
          ID = 0          (same as in Access-Request)
          Length = 68
          Attributes:
              Reply-Message = "bbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbb"
              State = {Magic Cookie to be returned along with user's
                       response; in this example 8 octets of data}

     [4] SIP proxy server to client:

          SIP/2.0 407 Proxy Authentication required
          From: sip:+19195551234@domain.com
          To: sip:+19195551212@cisco.com
          Call-ID: 12345600@cisco.com
          CSeq: 1 INVITE
          Proxy-Authenticate: CHAP-Password
            ;username="byerly"
            ;algorithm="MD5"
            ;id=0
            ;nonce="cccccccccccccccccccccccccccccccc"
          State: {Magic Cookie from Access-Challenge packet, unchanged}
                 (formatted as hex digits)

NOTE: In this instance, the RADIUS server chooses to re-challenge
      the SIP client with a new nonce.

     [5] SIP Client to SIP proxy server:

          INVITE sip:+19195551212@cisco.com SIP/2.0
          From: sip:+19195551234@domain.com
          To: sip:+19195551212@cisco.com
          Call-ID: 12345600@cisco.com
          CSeq: 2 INVITE
          Content-Type: application/sdp
          Proxy-Authorization: CHAP-Password
            ;username="byerly"
            ;algorithm="MD5"
            ;id=0
            ;nonce="cccccccccccccccccccccccccccccccc"
            ;response="dddddddddddddddddddddddddddddddd"
          State: {Magic Cookie from Access-Challenge packet, unchanged}
                 (formatted as hex digits)

Byerly/Williams      draft-byerly-sip-radius-00.txt              Page 10

Internet Draft       SIP Security using CHAP-Password         March 2000

     [6] SIP proxy server to RADIUS server:

          Code = 1        (Access-Request)
          ID = 1          (Note that this changes)
          Length = 71
          Request Authenticator = {NEW 16 octet CHAP challenge
                                   ()}
          Attributes:
              User-Name = "byerly"
              CHAP-Password = {1 octet CHAP ID followed by 16 octet
                               CHAP response
                               (dddddddddddddddddddddddddddddddd)}
              NAS-IP-Address = 192.168.1.16
              NAS-Port = 5
              State = {Magic Cookie from Access-Challenge packet,
                       unchanged}

     [7] RADIUS server to SIP proxy server:

          Code = 2        (Access-Accept)
          ID = 1          (same as in Access-Request)
          Length = 30

     [8] SIP proxy server to next hop UAS:

          INVITE sip:+19195551212@cisco.com SIP/2.0
          From: sip:+19195551234@domain.com
          To: sip:+19195551212@cisco.com
          Call-ID: 12345600@cisco.com
          CSeq: 2 INVITE
          Content-Type: application/sdp


9.2 User Authentication using backend RADIUS server -
    Without Server Challenge.

     [a] SIP Client to SIP proxy server:

          INVITE sip:+19195551212@cisco.com SIP/2.0
          From: sip:+19195551234@domain.com
          To: sip:+19195551212@cisco.com
          Call-ID: 12345601@cisco.com
          CSeq: 3 INVITE
          Proxy-Authorization: CHAP-Password
            ;username="byerly"
            ;algorithm="MD5"
            ;id=0
            ;nonce="eeeeeeeeeeeeeeeeeeeeeeeeeeeeeeee"
            ;response="ffffffffffffffffffffffffffffffff"
          Content-Type: application/sdp



Byerly/Williams      draft-byerly-sip-radius-00.txt              Page 11

Internet Draft       SIP Security using CHAP-Password         March 2000

     [b] SIP proxy server to RADIUS server:

          Code = 1        (Access-Request)
          ID = 0
          Length = 71
          Request Authenticator = {16 octet random number also used as
                                   CHAP challenge
                                   (eeeeeeeeeeeeeeeeeeeeeeeeeeeeeeee)}
          Attributes:
              User-Name = "byerly"
              CHAP-Password = {1 octet CHAP ID (00) followed by 16 octet
                               CHAP response
                               (ffffffffffffffffffffffffffffffff)}
              NAS-IP-Address = 192.168.1.16
              NAS-Port = 5

     [c] RADIUS server to SIP proxy server:

          Code = 2        (Access-Accept)
          ID = 1          (same as in Access-Request)
          Length = 56

     [d] SIP proxy server to next hop UAS:

          INVITE sip:+19195551212@cisco.com SIP/2.0
          From: sip:+19195551234@domain.com
          To: sip:+19195551212@cisco.com
          Call-ID: 12345601@cisco.com
          CSeq: 3 INVITE
          Content-Type: application/sdp

     There are two cases when a SIP client could pre-send a
     Proxy-Authorization that the RADIUS server might accept:
     1) The RADIUS server originally generated the nonce
        when challenging the SIP client on a previous call.
        The SIP client is reusing the previously sucessful
        Authorization for a new call.
     2) The SIP client originally generated the nonce.
        The parsed format of the nonce is known to both the SIP
        client and the RADIUS server.  The nonce contains a timestamp
        which the RADIUS server can extract and use to limit the
        replay window.  Since a RADIUS server silently discards
        invalid/unauthorized requests, this scheme is not subject
        to the form of the man-in-the-middle attack where Mallory
        sends a bogus request to the server and uses the response
        to make the client believe she is a legitimate server.

     To dos:
     1) Fix the RADIUS Lengths to be correct
     2) Calculate real MD5 hashes



Byerly/Williams      draft-byerly-sip-radius-00.txt              Page 12

Internet Draft       SIP Security using CHAP-Password         March 2000

     Outstanding issues:
     1) Do we/how do we support the RADIUS State: attribute?
        What are the implications for collision with (DCS-)State:
        object?
     2) Do we reuse the RADIUS NAS-Port and NAS-Port-Type attributes to
        allow the RADIUS server to have media port control over calls?
        (eg. using UDP port numbers for RTP streams)


10 Acknowledgements

   We would like to thank Roger Levesque, David Oran, Mike Thomas,
   David Daiker, Shail Bhatnagar, and Denise Caballero-McCann for
   discussions on the need for and improvements to this draft.
   We would also like to thank Tyrone Floryanzia for his insights on
   H.323 gateway/gatekeeper call authorization using RADIUS.

11 References

[SIP]  M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, SIP:
       Session Initiation Protocol", RFC 2543, March 1999.

[RAD]  C. Rigney, A. Rubens, W. Simpson, and S. Willens, "Remote
       Authentication Dial in User Service (RADIUS)," RFC 2138,
       April 1997.

[DIG]  Franks, J, et al. "HTTP Authentication: Basic and Digest Access
       Authentication," RFC 2617, June 1999.

[CHAP] Simpson, W.  "PPP Challenge Handshake Authentication Protocol
       (CHAP)", RFC 1994, August 1996.

[PAP]  Lloyd, B, W. Simpson. "PPP Authentication Protocols",
       RFC 1334, October 1992.

[PPP]  Simpson, W., Editor, "The Point-to-Point Protocol (PPP)", STD
       51, RFC 1661, DayDreamer, July 1994.

[MD5]  Rivest, R., and S. Dusse, "The MD5 Message-Digest Algorithm",
       MIT Laboratory for Computer Science and RSA Data Security,
       Inc., RFC 1321, April 1992.

[REQ]  Bradner, S., "Key words for use in RFCs to Indicate Requirement
       Levels," RFC-2119, March 1997.











Byerly/Williams      draft-byerly-sip-radius-00.txt              Page 13

Internet Draft       SIP Security using CHAP-Password         March 2000


Authors' Addresses

   Bryan J. Byerly
   Cisco Systems
   7025 Kit Creek Road
   P.O. Box 14987
   Research Triangle Park, NC 27709
   USA
   Email: byerly@cisco.com

   David Williams
   Cisco Systems
   7025 Kit Creek Road
   P.O. Box 14987
   Research Triangle Park, NC 27709
   USA
   Email: dwilli@cisco.com




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From sip-admin@lists.bell-labs.com  Mon May  8 23:39:37 2000
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I've just posted an Internet Draft on the compromise position
reached regarding the DCS group's two-stage invite proposal.
It's entitled "Integration of Resource Management and SIP for
IP Telephony", and is filed as draft-manyfolks-sip-resource-00.

Until it appears in the archives, it may be obtained via
anonymous ftp
ftp://ftp.research.att.com/dist/wtm/draft-manyfolks-sip-resource-00.txt

This draft is a combination of two previous drafts,
draft-ietf-mmusic-sdp-qos-00, and draft-dcsgroup-sip-resource-00,
and discusses how network QoS and security establishment can be
made a precondition to sessions initiated by SIP, and how the
resulting call setup mechanism achieves many goals, including
those of the PacketCable DCS project.

We anticipate no problems in reissuing this draft shortly after
Adelaide with "full conformance with Section 10 of RFC2026."

Also just posted are updates to 5 of the DCS group's Internet Drafts,
draft-dcsgroup-sip-arch-01
draft-dcsgroup-sip-call-auth-01
draft-dcsgroup-sip-state-01
draft-dcsgroup-sip-privacy-01
draft-dcsgroup-sip-proxy-proxy-01

These drafts reflect the discussion at the November working
group meeting, and further work since.

Until they appear in the archives, they may be obtained
via anonymous ftp
ftp://ftp.research.att.com/dist/wtm/draft-dcsgroup-sip-arch-01.txt
ftp://ftp.research.att.com/dist/wtm/draft-dcsgroup-sip-call-auth-01.txt
ftp://ftp.research.att.com/dist/wtm/draft-dcsgroup-sip-state-01.txt
ftp://ftp.research.att.com/dist/wtm/draft-dcsgroup-sip-privacy-01.txt
ftp://ftp.research.att.com/dist/wtm/draft-dcsgroup-sip-proxy-proxy-01.txt

Bill Marshall
wtm@research.att.com




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From: "Neville O'reilly" <noreilly@netspeak.com>
To: "'S_S_Dinesh/India/IBM@au1.ibm.com'" <S_S_Dinesh/India/IBM@au1.ibm.com>,
        Shail Bhatnagar <shbhatna@cisco.com>
Cc: sip@lists.research.bell-labs.com
Subject: RE: [SIP] Re: "headers" in SIP-URL
Date: Mon, 8 May 2000 15:06:36 -0400 
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I believe that's true i.e. most of the extensions (maddr,headers, etc) are
not allowed in the To, From parameters, but can be used in the Contact
parameters 

-----Original Message-----
From: S_S_Dinesh/India/IBM@au1.ibm.com
[mailto:S_S_Dinesh/India/IBM@au1.ibm.com]
Sent: Monday, May 08, 2000 4:40 AM
To: Shail Bhatnagar
Cc: sip@lists.research.bell-labs.com
Subject: [SIP] Re: "headers" in SIP-URL




Shail Bhatnagar wrote:
>
> A look at the syntax of SIP-URL shows that headers can be defined in a
SIP-URL
> itself. However, the spec also says they MUST NOT be present in From/To
and
> Request-URI. Does it mean they can be present in Contact header (the
addr-spec
> portion of Contact) ??
> I don't see any good reason for headers being allowed inside a SIP-URL
and I
> would be curious to know how many implementations actually introduce it
in
> SIP-URL ( probably the Contact header) and rely on it.

In a redirect, you can play tricks such as adding new headers to the
next, redirected call. (For example, the redirect server could provide
the password for another call.) While I doubt that it is used right now,
it seems like a feature worth having around.


>
> Thanks,
>
> --
> Best regards,
> Shail

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




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Subject: [SIP] Re: About "The SIP servlet API"
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Hi,

The "SIP Servelet API" and "Building a SIP Stack" are two very different
things.

If you want to build a SIP Stack, which can be used in many softwares as
you say, you are looking at a generic stack that can
be used to build UAs, Proxies, Redirects etc. In which case, you should not
be bulding a SIP Servelet API only - this document models itself to the
usual Web Server-  Sevelet interface and focusses prmarily on Java, and for
a specific intention - deferring SIP controls to Java classes as 'SIP
Servelets'

Again, thats harldy building a stack to suit all applications.
Of course, you could derive APIs from it if you so wish, but the bottom
line is what you want to do (build a generic sip stack) and what the
servelet API propses are two different things.

If your friend wants a servelet interface, and you think the servelet API
draft suits you, sure you could implement it, but again, it does
not *replace* a generic SIP stack, it would add on to it.

Why dont you look at
http://www.cs.columbia.edu/~hgs/sip/implementations.html   - there are
companies that have implemented generic stacks, and some of their links
(follow it from the implementations page) give you an idea of expected
features from a generic SIP Stack


Regds
Arjun

--
Arjun Roychowdhury @ Hughes Software Systems










"yinshaohua" <yin@huawei.com.cn> on 03/14/2000 01:02:09 PM

Please respond to "yinshaohua" <yin@huawei.com.cn>

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

Subject:  About "The SIP servlet API"




Hi :
    Now I have a discussion with my fellow, would you give some suggestions
?
    We want to make a SIP Stack which can be used in many softwares. My
fellow consider that the SIP Stack will be realized providing that it
presents the API of the draft " The SIP Servlet API". But I have some
questions about the API:
    A. When using in sip server,the servlet dispaly less role: decode the
SIP message and the decoded header can be reached by server , store the
trasaction (call-id, to etc), create
message and may set some values according to the
transaction,retransmissions. In section 5.3 of the draft, there writes "
These headers (call-id, from,to,cseq) should not need to be manipulated
directly by servlets ",  I think A SIP stack SHOULD complete essential sip
call management and process some parameters, sending only the useful
infomation for service but not all parameters of message to other module.
    B. As a module of a software, the provided API is so many that it can
be
said the module is not encapsulated. So it not benefit to software
maintenance. I think a SIP stack only gives a series of abstract API which
can be used by other module to complete the service stipulated by the
protocol
    C. Due to so many API, it is difficult to replant a servlet to other
use, meanwhile, the
servlet is diffcult to maintain because it have not a legible limit with
other module.
    D. The SIP STACK should guarantee the SIP message sent by it accord
with
SIP protocol, but the servlet will not due to offering the API.So there may
send unlawful message when the servlet use in other software.
    Please give some suggestions. Thanks!

Shaohua Yin

2000/3/10













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S_S_Dinesh/India/IBM@au1.ibm.com wrote:

> Folks,
>
> I've just posted an individual Internet Draft which may be of interest.
> Its entitled "Guidelines for Authors of SIP Extensions". Its discusses
> issues that writers of SIP extensions should consider while developing
> them and writing them up. It also gives some guidance on what things
> make good SIP extensions, and what things do not. Until it appears in
> the archives, you can pick up a copy at:
>
> http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-sip-guidelines-00.txt
>
> Thanks,
> Jonathan R.
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

CAN'T THESE BE SHUT OFF ? They all seem to be old, and E-mail to the source bounces.



--
                                 Regards
                                 Marshall Eubanks


T.M. Eubanks
Multicast Technologies, Inc
10301 Democracy Lane, Suite 410
Fairfax, Virginia 22030
Phone : 703-293-9624
Fax     : 703-293-9609
e-mail : tme@21rst-century.com




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> John Mainwaring wrote:
>
> |The problem is related to past discussions on the SIP mailing list
> |regarding overlap dialing. A terminating gateway would look at
> |sip:1212344, and know how to route it, but might not know whether
> this
> |was a dialed number with more digits coming, or an LRN. Does it
> matter?
> |Not being an ISUP expert, I'm not sure. If it does matter, something
> |like user=lrn within the SIP URL would fix that.
>
> user=lrn is almost right.  If you have queried a portable number and
> found that it is not ported, then the DN has the same status as an LRN
> would have if the number were ported (i.e. it should be treated as
> authoritative for routing).  user=np-queried might be closer to the
> mark.

I wonder if we can't think about this problem in more general terms.

In the SIP model, each proxy owns some piece of the namespace
(engineering.dynamicsoft.com, for example), and is therefore responsible
for the definition of the user space (jdrosen@) within that domain.
Thats why each proxy can, an usually will, perform some kind of URI
translation independent from other proxies. Our underlying assumption
was that usernames only have meaning within their domain. For example,
with sip:jdrosen@dynamicsoft.com, only a dynamicsoft.com proxy knows how
to handle jdrosen. There might be jdrosen's in other domains that aren't
me. However, in the case of telephone numbers, the proxy that owns the
domain does not necessarily own the user space. sip:5551212@gateway.com,
for example, uses a user name that is not "owned" by gateway.com, in the
sense that this name has valid semantics outside of gateway.com. Other
domains, like wcom.com, would know how to handle this number as well.
Its possible that we might even see things besides phone numbers as user
names that are valid across multiple domains. An example is (heaven
forbid), social security numbers (in the US, at least):

sip:123-45-6789@wcom.com;user=ssn

Now, I might imagine that some kind of mobility services might be
supplied for these names, so they might need to be translated also.

I'll call these type of user names "cross domain user names".

With cross domain user names, this is where we run into issues like "has
the number been queried yet". However, I think we might see other
translation mechanisms besides LNP. Soooo, might it not be useful to
define URI parameters that indicate what global translation services
have been applied to a cross domain user name?

As an example:

sip:15551212@cisco.com;user=phone;trans="lnp,gdbs"

where lnp and gdbs are some kind of global translation services that
apply to names of the type phone.

For cross domain user names, we will always need a URI parameter to
indicate the global namespace from which these names are drawn. In the
case of phone numbers, thats user=phone. Our original intent with the
user=phone parameter was to specify only the type of BNF used for the
user part of the URI. But, the more general operation of specifying both
the format and the namespace seems useful.


>
> It's also necessary to define what the TRIP should do differently
> depending on whether the np-queried indication is present.  I must
> admit I came into this discussion mainly because I have worked on
> LNP.  I assume the objective is not to route unqueried calls to the
> donor proxy.

Hmm. This is something thats also different from the PSTN. The cost of
routing to the donor proxy in SIP is much less, since proxies can be
stateless. Thus, if a proxy receives a call for some number that is
ported, it stateless proxies the call towards the next destination. The
result is that there is no triangle routing for media (there generally
isn't), nor for subsequent signaling. There is only marginal cost for
the donor provider. There are obviously polticial and business issues;
one may not want to trust or rely upon the donor proxy at all, as is the
case of LNP in the US. But, the general SIP model is that we assume
proxies do route calls; no security measures can prevent a proxy from
dropping messages. Why, in the case of LNP, do we now need to assume
these proxies won't route calls?




 I have noted that originator query may not be practical
> for nationwide calling and that donor proxy is too late unless you're
> prepared to consider RTP.

The assumption of the impractability of originator queries has
everything to do with the existing structure of LNP. Its mainly because,
AFAIK, the systems are different all over the world. However, if we use
the IP model, with a DNS solution, its no harder to query for a number
on the other side of the planet as it would be for a number next door.

Also, pardon my ignorance, but what is RTP? Many of us on this list know
RTP as the Real Time Transport Protocol (rfc1889) but I don't think
thats what you mean.


 One answer might be for the TRIP to provide
> a response on unqueried numbers that routes to an intermediate server
> that does know how to do NP queries for the destination DN.

TRIP is not a query protocol. Its a routing protocol. This would be a
SIP function, and is already possible. "408 Not Found".

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

---------
This message came from the IETF IPTEL Working Group Mailing List.




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Alan Johnston wrote:
>
> Rohan,
>
> You are correct - the Session:Media header is missing from all 183
responses in
> the document.  That will be fixed - thanks for pointing this out.
>
> I think the sequence of PRACKs and 200 OKs in scenario 4.1.2 (page 84) is
> correct.  PRACK is an end-to-end not hop-by-hop method (see Section 3 of
> draft-ietf-sip-100rel), so the 200 OK response to it is generated by GW 1
and
> forwarded by Proxy 1.  This is why there is no requirement that proxies
support
> the reliable response feature, but both User Agents must support it.

Correct.

-Jonathan R.

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




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To: IETF SIP <sip@lists.research.bell-labs.com>
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Joerg Ott <jo@tzi.uni-bremen.de>
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Subject: [SIP] Draft agenda SIP WG published
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The draft agenda for Adelaide is at

http://www.softarmor.com/sipwg/meets/IETF47/agenda.html

with HTML attached for those of you who are web-deprived.

If you see any glaring errors, please let Jonathan, Joerg, and I know ASAP.
It's due at noon later today. I hope most of my errors were caught in my
requests for feedback on the agenda request list.

--
Dean Willis
SIP WG cochair

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Subject: [SIP] Informal Last Call: Common Gateway Interface for SIP
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With the submission of the most recent version of the SIP Common Gateway
Interface, we'd like to announce an informal Last Call on this document (in
place of a Working Group last call, since this draft has no associated
working group).  We'll solicit comments for four weeks, until March 31,
2000.  At that time we'll submit this draft for Informational RFC status.

Comments on the draft should be sent to the SIP mailing list
sip@lists.research.bell-labs.com, or to the authors.



     Title          : Common Gateway Interface for SIP
     Author(s) : J. Lennox, J. Rosenberg, H. Schulzrinne
     Filename  : draft-lennox-sip-cgi-03.txt,.ps
     Pages          : 38
     Date      : 02-Mar-00

In Internet telephony, there must be a means by which new services
are created and deployed rapidly. In the World Wide Web, the Common
Gateway Interface (CGI) has served as popular means towards
programming web services. Due to the similarities between the Session
Initiation Protocol (SIP) and the Hyper Text Transfer Protocol
(HTTP), CGI seems a good candidate for service creation in a SIP
environment. This draft proposes a SIP-CGI interface for providing
SIP services on a SIP server.

URLs for this Internet-Draft are:
http://www.ietf.org/internet-drafts/draft-lennox-sip-cgi-03.txt
http://www.ietf.org/internet-drafts/draft-lennox-sip-cgi-03.ps
http://www.cs.columbia.edu/~lennox/draft-lennox-sip-cgi-03.pdf

The Postscript and PDF versions contain changebars indicating changes from
the previous version.

--
Jonathan Lennox
lennox@cs.columbia.edu

---------
This message came from the IETF IPTEL Working Group Mailing List.




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archow@hss.hns.com wrote:
>
> Hi,
> We need to use SIP to connect to PSTN side Fax, click to dial etc.
> I read through the PINT draft which is supposeldy catering to this type
of
> service.
>
> However, I wanted to know which approach to select. Is PINT the standard
> way for PSTN service interaction ?
> It seems whatever PINT states can be done with regular SIP - or is that
> incorrect

PINT provides three specific functions - click-to-dial, click-to-fax,
and click-for-content. It is not a general purpose protocol for PSTN
service interaction. The scope of PSTN service interaction is
sufficiently large that I think there will be many ways. For example,
look at the recent SPIRITS working group.

It is possible to do these three services with vanilla SIP. There are
significant differences in how the service is delivered, though, with
pros and cons for both approaches. A colleague and I submitted a draft
on Friday entitled "Third Party Call Control in SIP", which looks at a
general purpose SIP approach for third party call control (its not an
extension, just a usage). One of the services discussed is click to
dial, and there is a brief comparison to pint. I'm sure there are other
ways of providing this service beyond pint and whats in the draft.

Until it appears in the archives, you can pick up a copy of the draft
at:
http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-sip-3pcc-00.txt

>
> And is my understanding that PINT is nothing but SIP/SDP  with some
> extensions and is analogous to the various SIP extensions drafts
> that are in existence today ?

Yes. PINT extends SDP more than SIP, since it describes sessions that
are purely on the telephone network.

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




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Subject: [SIP] Re: SIP & RTSP interworking?
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Linden deCarmo wrote:

> I like this a lot better, but the problem I have is this, we'd like for
BOTH
> sessions to be active at the same time and both of your proposals require
a
> teardown of the SIP session.
>

Linden,

I still dont understand what the application is. Your first note said :

*************
What I'd like to do is setup a session with SIP (via an INVITE), then
control streaming aspects via RTSP (i.e. play, pause, stop etc.).
*************

Implying to me that there was only one session.

Your second note says:

**************
I like this a lot better, but the problem I have is this, we'd
like for BOTH
sessions to be active at the same time and both of your proposals
require a
teardown of the SIP session.
**************

Implying that there were two. (which I think makes sense).

I think there are broadly two scenarios:

a)  Sending an RTSP URL to a SIP UA as part of redirection or call
termination
or transfer. The SIP session in question would be terminated.  From the SIP
point of view, it seems to make sense to put the RTSP URL in a Contact: or
Also:, exact callflows and details to be worked out.

b) Have a RTSP controlled stream as part of a conference ie an addon to an
existing or about-to-exist SIP session.
There could be a number of interesting variants to this.

One of them that follows the "SIP-UA knows RTSP" model:
You can have one SIP UA be the RTSP client (ie the owner/controller of the
stream if you will), and direct the stream to each SIP UA in the
session(could
be via multicast as well). Other UAs dont need to know anything about RTSP.
In
which case SIP needs a way to inform UAs that  this new RTP stream will be
added
to the session which I guess it should be able to do using re-INVITEs.
If you go with the model that the RTSP server understands SIP, then RTSP
provides a means to pass the conference ID to the RTSP server so that it
can
join the SIP session and link subsequent RTSP operations to the SIP
session.
Such IDs are opaque to RTSP protocol , so the usage would depend on the
server
and application in question.

-Anup.

> Linden deCarmo
> Netspeak Corporation
> 902 Clint Moore Road
> Suite 104
> Boca Raton, FL 33487
>
> > -----Original Message-----
> > From: Kundan Singh [SMTP:kns10@cs.columbia.edu]
> > Sent: Wednesday, March 15, 2000 3:10 PM
> > To:   Linden deCarmo
> > Cc:   'Dean Willis'; sip@lists.research.bell-labs.com
> > Subject:      RE: SIP & RTSP interworking?
> >
> > --
> > Kundan Singh     http://www.cs.columbia.edu/~kns10
> >
> > > This is an interesting suggestion, but the scenario I'm operating
under
> > is
> > > this:
> > >
> > > UAC<->UAC (in this case the UAC really is an RTSP server), both must
> > > understand SIP and RTSP.  I need a way to tie the sessions between
two
> > > distinct protocols.  Is a viable alternative to embed the RTSP SETUP
> > message
> > > as a SIP body for the INVITE?  This would clearly associate the
> > sessions,
> > > but I hate to have a proprietary solution that wouldn't work with
other
> > > products that understood SIP & RTSP.
> >
> > In that case, the UA may return the RTSP Contact: URI in (say) 3xx
> > response.
> >
> >  UA1          UA2
> >      INVITE
> >  -------------->
> >
> >    3xx Moved
> >    Contact:rtsp://server/resource
> >  <-------------
> >
> >    SETUP rtsp://server/resource
> >  -------------->
> >  . . .
> >
> > Since UA here understand both SIP and RTSP it should not be
> > a problem.
> >
> > If the SIP call is already established, and UA2 wants to
> > switch to RTSP,  then (IMO) it may use blind call transfer
> > (BYE,Also:rtsp://...) to ask UA1 to connect to RTSP
> > server of UA2. Comments ??
> >





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Jonathan,

let me say that I' m a bit tired  but, I'll try anyway... (forgive me if I
say anything terribly
wrong)

Let' s even assume I can put each terminal behind a firewall, as you
described (which is
something that, in our architecture, might even be feasible):

+---+               +------------+
| te1 |--------->|FW+Proxy1|------->+
+--+        +------------+            |
                                                           |
                                                           |
+---+               +------------+            |
| te2 |<---------|FW+Proxy2 |<-------+
+--+        +------------+


Now, by assumption, TE1 is hacked: I can do whatever I want with it.

Well this network must be connected to the Internet: let's assume there is
a firewall3 towards
the Net.
It is configured to let HTTP pass through as well as incoming INVITE and
ACK messages (this is
necessary to interoperate with the other PNOs).

Now let me start my own business "freecalls Inc." (or, at least, "cut price
calls inc...."), also
selling  terminals that

   *  incapsulate SIP in HTTP ===> [FW+Proxy1] doesn't detect the INVITE
nor the ACK
   *  send the packetized voice via TCP to my magic server in the Net

+---+               +------------+
| te1 |--------->|FW+Proxy1|--------+
+--+        +------------+            |
                                                           |+-----+
+----------------------------+
                                                           | FW3
|--------->|Server (in the
public Internet)+
                             +-----+       +----------------------------+
                                                          |
                                                           |
+---+               +------------+            |
| te2 |<---------|FW+Proxy2 |--------+
+--+        +------------+


and that' s  it: the server performs all the required translatons and sends
all the SIP
signalling and an RTP stream to TE2.

Well, I have to admit it is not very optimized, but I designed it in half
an hour....
.....I' sure our crowds of estimed hackers will be able to devise some much
more efficient
solutions....

Ciao
Giuseppe

P.S.
1st optimization: you said that "This, of course, forces you to provide
application-level
gateways for every UDP-using protocol your customers want to use."
Instead of tunneling Firewall1 with HTTP+TCP, I could try with one of such
UDP-based protocols...
(much better !)









Jonathan Lennox wrote:

> On Tue, March 28 2000, "Giuseppe Ricagni" wrote to
"sip@lists.research.bell-labs.com" saying:
>
> > during these days in Adelaide, I have already had the chance to discuss
> > with some well known SIP experts about this topic, nevertheless I would
> > like to pose the same question to the list: maybe someone has some
other
> > idea.
> >
> > Let' s assme that the scenaro is:
> >
> >    * there are a number of "standard" SIP terminals, whose software is
> >      not modifiable by the user (nor any other application can be
> >      installed)
> >    * such SIP softare forces SIP to work in proxy mode
> >    * the proxy actually bills the calls
> >
> > Now let' s introduce a "hacked" terminal, and lets' s say that the
> > software on such terminal is completely free, in the sense that
whatever
> > is needed we can modify it and install it.
> >
> > The question is: is it posible to define, to any extent, a mechanism
> > capable of preventing the "hacked" terminal to bypass the proxy and
call
> > the "standard" endpoints witout being billed ?
> >
> > Please CC replies to gricagni@tin.it (it is easier for me to read email
> > from such mailbox here in Adelaide)
>
> Put the terminals behind a firewall which doesn't allow random UDP
packets
> through.  Have the SIP proxy tell the firewall to open up the ports for
each
> call's RTP streams after it processes the corresponding INVITE.  Close
them
> when you see the BYE or when a session-timer expires (you'll need session
> timers for this to work).
>
> This, of course, forces you to provide application-level gateways for
every
> UDP-using protocol your customers want to use.  It also leads to the
> question of why your customers won't just decamp to a provider who gives
> them proper end-to-end IP service, and only bills for true value-added
> features.
>
> --
> Jonathan Lennox
> lennox@cs.columbia.edu

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S_S_Dinesh

I just got this email (this morning) even though it likely was sent in the past
(few months old I think) -- so forgive the following please if this isn't a new
position. But in case this was just sent, I have the following response:

I'm confused (again), from the line below "...we make protocols that run on the
Internet, not the PSTN..." Isn't kinda obvious IP Protocols don't run on a TDM
infrastructure where there isn't even IP?

SIP is a Protocol that runs on IP, right?

So why does this response below want to limit SIP to a Public Internet? Is
MCI/WorldCom's Carrier-grade VoIP Network they're rolling out now considered
the 'Public Internet'? I kinda don't think so Anything Public doesn't have a
fee for services -- which I have to believe MCI/WorldCom plans on charging for
use of here.

AT&T's Broadband IP Voice network will also have a fee for service -- is that
the 'Public Internet'? I kinda don't think so again.

Will there be QoS to support SIP on what you consider the 'Public Internet'?
That's a good question. Cause if there isn't -- there likely isn't SIP.

So.... SIP isn't just for the Public Internet. SIP is for where IP exists that
can support a high QoS infrastructure and application that can run on that
infrastructure, agreed? If people don't agree with this -- I'd love to hear
where they plan on limiting SIP deployments.

So.... if SIP is for infrastructures that can support SIP -- why is this only
the Internet? S_S_Dinesh, are you only connected to a Internet? Most of us
aren't. We're connected to Private Networks that connect to the Public
Internet. Why should SIP only start between company networks?

 From another point of view -- if you, the ones that have only Public Internet
Access (via DSL or Cable) and have SIP capability from that location, want to
initiate a SIP session between yourself and someone that isn't in a Public
Internet connected location, that session doesn't work does it? Based on what
you've decreed, it cannot.

What if you were in fact initiating a session to someone's IP Phone that ran
SIP in some internal network that was connected to that Public Internet you
were? Could that session exist? If it can, my I-D can too.

If you're objecting to my extension to the Priority Header-Field, were you also
opposed to what is there now? There are 4 levels -- I'm merely suggesting a
5th. Jonathan admitted at Adelaide that ".. no one uses this now.." Great.
Except when he asked the audience (of substantial size) what they felt about it
-- it appears nearly unanimous that this was worth pursuing.

I agree I need to rewrite the I-D towards what only involves SIP (in which
Preemption is merely referenced and not mandated within it). But if there is to
be Priority within a SIP network, semantics for how and examples of where,
should be written and accepted by this WG.

BTW -- if will be actually changed within the SIP work is the addition of a
single value to an existing Header-Field, why are you so opposed to this work?
If it isn't used by you -- so what? If it is used, and in this case it is
mandatory for certain installations of very large size -- what harm did it
cause you? It should be little if any: an unused Header-Field with an
additional value.

What parts of this don't make sense to people?


At 02:10 PM 5/8/2000 +0530, S_S_Dinesh/India/IBM@au1.ibm.com wrote:
>
>
>
>
>Ken Carlberg wrote:
>>
>> > On your final thread, it should certainly be possible to define
>> > an extension to SIP that is not useful in the PSTN; extensions
>> > that are useful only within an enterprise for example should
>> > be acceptable, or in the home, as well as something like MLPP
>> > which is useful within a government communications system.
>>
>> Agreed with respect to something that may or may not be "useful".  But
>the
>> kicker that I'm curious about is whether the proposed specification is
>> illegal (not allowed) when applied to the general public.
>
>Here at IETF, we make protocols that run on the Internet, not the PSTN.
>As there are no regulations regarding this kind of thing on the
>Internet, it is not an issue for us.
>
>-Jonathan R.
>
>
>--
>Jonathan D. Rosenberg                       72 Eagle Rock Ave.
>Chief Scientist                             First Floor
>dynamicsoft                                 East Hanover, NJ 07936
>jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
>http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
>http://www.dynamicsoft.com
>
>
>

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

James M. Polk
Sr. Product Manager, Multiservice Architecture and Standards
Enterprise Voice Business Unit
Cisco Systems
Dallas, Texas
w) 972.813.5208
f)  972.813.5280
www.cisco.com
--=====================_64424256==_.ALT
Content-Type: text/html; charset="us-ascii"

<html><div>S_S_Dinesh</div>
<br>
<div>I just got this email (this morning) even though it likely was sent
in the past (few months old I think) -- so forgive the following please
if this isn't a new position. But in case this was just sent, I have the
following response:</div>
<br>
<div>I'm confused (again), from the line below &quot;...we make protocols
that run on the Internet, not the PSTN...&quot; Isn't kinda obvious IP
Protocols don't run on a TDM infrastructure where there isn't even
IP?</div>
<br>
<div>SIP is a Protocol that runs on IP, right?</div>
<br>
<div>So why does this response below want to limit SIP to a Public
Internet? Is MCI/WorldCom's Carrier-grade VoIP Network they're rolling
out now considered the 'Public Internet'? I kinda don't think so Anything
Public doesn't have a fee for services -- which I have to believe
MCI/WorldCom plans on charging for use of here.</div>
<br>
<div>AT&amp;T's Broadband IP Voice network will also have a fee for
service -- is that the 'Public Internet'? I kinda don't think so
again.</div>
<br>
<div>Will there be QoS to support SIP on what you consider the 'Public
Internet'? That's a good question. Cause if there isn't -- there likely
isn't SIP.</div>
<br>
<div>So.... SIP isn't just for the Public Internet. SIP is for where IP
exists that can support a high QoS infrastructure and application that
can run on that infrastructure, agreed? If people don't agree with this
-- I'd love to hear where they plan on limiting SIP deployments.</div>
<br>
<div>So.... if SIP is for infrastructures that can support SIP -- why is
this only the Internet? S_S_Dinesh, are you only connected to a Internet?
Most of us aren't. We're connected to Private Networks that connect to
the Public Internet. Why should SIP only start between company
networks?</div>
<br>
<div> From another point of view -- if you, the ones that have only
Public Internet Access (via DSL or Cable) and have SIP capability from
that location, want to initiate a SIP session between yourself and
someone that isn't in a Public Internet connected location, that session
doesn't work does it? Based on what you've decreed, it cannot.</div>
<br>
<div>What if you were in fact initiating a session to someone's IP Phone
that ran SIP in some internal network that was connected to that Public
Internet you were? Could that session exist? If it can, my I-D can
too.</div>
<br>
<div>If you're objecting to my extension to the Priority Header-Field,
were you also opposed to what is there now? There are 4 levels -- I'm
merely suggesting a 5th. Jonathan admitted at Adelaide that &quot;.. no
one uses this now..&quot; Great. Except when he asked the audience (of
substantial size) what they felt about it -- it appears nearly unanimous
that this was worth pursuing.</div>
<br>
<div>I agree I need to rewrite the I-D towards what only involves SIP (in
which Preemption is merely referenced and not mandated within it). But if
there is to be Priority within a SIP network, semantics for how and
examples of where, should be written and accepted by this WG.</div>
<br>
<div>BTW -- if will be actually changed within the SIP work is the
addition of a single value to an existing Header-Field, why are you so
opposed to this work? If it isn't used by you -- so what? If it is used,
and in this case it is mandatory for certain installations of very large
size -- what harm did it cause you? It should be little if any: an unused
Header-Field with an additional value.</div>
<br>
<div>What parts of this don't make sense to people?</div>
<br>
<br>
<div>At 02:10 PM 5/8/2000 +0530, S_S_Dinesh/India/IBM@au1.ibm.com
wrote:</div>
<div>&gt;</div>
<div>&gt;</div>
<div>&gt;</div>
<div>&gt;</div>
<div>&gt;Ken Carlberg wrote:</div>
<div>&gt;&gt;</div>
<div>&gt;&gt; &gt; On your final thread, it should certainly be possible
to define</div>
<div>&gt;&gt; &gt; an extension to SIP that is not useful in the PSTN;
extensions</div>
<div>&gt;&gt; &gt; that are useful only within an enterprise for example
should</div>
<div>&gt;&gt; &gt; be acceptable, or in the home, as well as something
like MLPP</div>
<div>&gt;&gt; &gt; which is useful within a government communications
system.</div>
<div>&gt;&gt;</div>
<div>&gt;&gt; Agreed with respect to something that may or may not be
&quot;useful&quot;.&nbsp; But</div>
<div>&gt;the</div>
<div>&gt;&gt; kicker that I'm curious about is whether the proposed
specification is</div>
<div>&gt;&gt; illegal (not allowed) when applied to the general
public.</div>
<div>&gt;</div>
<div>&gt;Here at IETF, we make protocols that run on the Internet, not
the PSTN.</div>
<div>&gt;As there are no regulations regarding this kind of thing on
the</div>
<div>&gt;Internet, it is not an issue for us.</div>
<div>&gt;</div>
<div>&gt;-Jonathan R.</div>
<div>&gt;</div>
<div>&gt;</div>
<div>&gt;--</div>
<div>&gt;Jonathan D.
Rosenberg&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
72 Eagle Rock Ave.</div>
<div>&gt;Chief
Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
First Floor</div>
<div>&gt;dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
East Hanover, NJ 07936</div>
<div>&gt;jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
FAX:&nbsp;&nbsp; (732) 741-4778</div>
<div>&gt;<a href="http://www.cs.columbia.edu/~jdrosen" EUDORA=AUTOURL>http://www.cs.columbia.edu/~jdrosen</a>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
PHONE: (732) 741-7244</div>
<div>&gt;<a href="http://www.dynamicsoft.com/" EUDORA=AUTOURL>http://www.dynamicsoft.com</a></div>
<div>&gt;</div>
<div>&gt;</div>
<div>&gt;</div>
<br>

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

--=====================_64424256==_.ALT--



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To: "Gazal, Elly" <Elly_Gazal@icomverse.com>
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Subject: [SIP] Re: Message Waiting Indication Supplementary Service
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Hi,

Getting or checking the status of a voice mailbox is orthogonal to SIP's
purpose (establishing sessions). You could use a separate protocol, or you
could use either the SIP INFO or SUBSCRIBE/NOTIFY methods to provide
supplemental information to the phone/endpoint.  You could use an XML,
plain text, or other body type to convey this information, or an Event:
header.

thanks,
-rohan


At 12:12 AM 3/20/00 , Gazal, Elly wrote:
>Can any one suggest how does Message Waiting Indication Supplementary
>Service is achieved by SIP?
>
>Regards,
>
>Elly.
>





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Hear hear !!

-----Original Message-----
From:	Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent:	Tuesday, May 09, 2000 3:20 AM
To:	ozsip@oz.com
Cc:	sip@lists.research.bell-labs.com
Subject:	Re: [SIP] Content-Language header

What you are describing is actually something different still.
Accept-Language describes the language that may  be used for the content
in a response. This has nothing to do with the language an operator you
might be connected to is able to speak; for that, the caller preferences
extension is targeted (in fact, it has preferences for languages
spoken). This is also different from Content-Language, which describes
the language of the html or text in a SIP response. As Henning has
pointed out, the only real use for this is some kind of automated
translation systems.

I know its optional and not hard, but I would rather not add more stuff
to an ever-more-complex protocol unless there is a very well defined
need. Since there is realistically no useful application right now of
Content-Language, I'd rather not add it.

-Jonathan R.

ozsip@oz.com wrote:
> 
> > Realistically, what are the applications for Content-Language? What
> > would a client do differently on receiving a Spanish version vs.
> > English? I see the value in Accept-Language, so that I get the right
> > thing back in the first place, but Content-Language?
> 
> Perhaps if you would send a SIP invitation with "Content-Language:
Spanish"
> to sip:support@somewhere.com, you would prefer the proxy at somewhere.com
> to route your call to a support representative who spoke spanish.
> 
> -David H. Brandt
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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It certainly should, and not only for BYE requests. This is discussed in
http://www.cs.columbia.edu/~hgs/sip/notes.html (see the note on
Record-Route towards the end of the document) and in
http://www.cs.columbia.edu/~hgs/sip/drafts/draft-ietf-sip-2543bis-00.pdf
(section 6.3.33).

---
Igor Slepchin


Sunitha Kumar wrote:
>
> It is given that the record route in 200 OK is copied into  the Route
> headers for subsequent requests from the caller.  My question was, if
> we needed to make the BYE from the *callee*  traverse the same set of
> proxies, then  should this be rememberd from the INVITE request
> obtained from the caller, or does the via field help in that.
>
> Thanks!
>
> --
> Sunitha Kumar
> Software Engineer
> Vovida Networks
> (408) 957 - 6374
>
>




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Henning Schulzrinne wrote:
>
> A few low-level comments:
>
> - I'm not sure "TRANSFER" is the right verb here, since transfer implies
> moving something from one place to another. All TRANSFER does is
> effectively ask the recipient to send another INVITE. It would be nice
> if the same action could be used (say) for mesh conferences.

It would be nice, and indeed this is the motivation which led us to use
Also for both in the original call control draft. Problem is, a full
mesh conference requires much more than just "send an INVITE to
so-and-so". It requires a fair amount of synchronization and additional
semantics, beyond what is needed for transfer.

However, I agree that the mechanism might have other uses besides
transfer, so something more functional would be nice. I also agree that
we should figure out the primitive functions we need to accomplish the
call control capabilities, and then specify extensions for the
primitives rather than the services. The services themselves can then be
described through informational documents that reference the extensions.
I think the idea proposed in the framework, of basically breaking call
control into separate pieces, is probably a good idea.


> - Using TRANSFER instead of INVITE with Also does have one major
> disadvantage, since the originator of the request will retransmit with
> T2 (4 seconds).

An excellent point. Another advantage of BYE/Also over a new method is
that it saves two messages (separate TRANSFER/response and BYE/response)
when implementing your basic blind transfer.

However, I think a new method is better, in the end. It becomes
increasingly complex to figure out how to handle an INVITE or BYE
request if it can be filled with tons of headers, each doing a separate
thing. Also, by overloading BYE, the interpretation of the response
becomes harder. If the response to a BYE/Also was 400, was this because
the transfer failed, or because the BYE itself was bad? Using a specific
method makes it clear what the response means. Also, using a new method
means that the same mechanism can support unattended transfers, blind
transfers, and a bunch of others.

So, if we go by the philosophy in the "Guidelines for Authors of SIP
Extensions" draft, generality and flexibility wins over efficiency,
which speaks for a new method.


>
> - According to the rules for feature names, IETF-based extensions are
> single words, to avoid any possible confusion with proprietary
> extensions having inverted domain names. Thus, it should probably just
> be Requires: transfer rather than cc.transfer. (On the other hand, it's
> not clear this is needed at all - we don't need Require for new methods,
> since they already have a mechanism to say "sorry, je ne comprend pas").

Require is definitely not needed. This is an advantage to new methods as
opposed to new headers in an existing method.

It would be really nice, however, to know ahead of time whether all
parties supported transfer (and any other mechanisms to be defined). The
Supported extension draft defines a way in which the UAC would indicate
to the UAS that it knows the new mechanism, but that draft does not
mandate that a UAS return a Supported header in the response to any
method (only OPTIONS). We can add this, and say that if a UAS receives a
request (with any method) with a Supported header, it should place a
Supported header in the response and list those features it knows. This
would allow a caller to know whether the called party supports transfer
before trying it. Thoughts?

I noted the following in the transfer draft:
> Explicit non-Requirements
>
>    1. There is no requirement for the Transfer Target to be notified
>    that he is receiving an INVITE as the result of a TRANSFER request
>    (see comment 2).

Why? I really liked the idea of knowing that this call was the result of
a transfer, and knowing who the transferor was. A UA can always ignore
this information, and then get the basic service you have today.


The document says TRANSFER can't have a body. Why not? I can think of
some cool uses for this down the road. Why rule it out?

The draft says:
> UA receiving a well-formed TRANSFER request SHOULD request approval
>    from the user to proceed. In the absence of that request, or upon
>    receiving approval from the user, the UA MUST submit an INVITE to
>    the resource identified by the Transfer-To: header using the Call-ID
>    from the TRANSFER request.

This thing about the Call-ID being copied from the TRANSFER seemed odd.
Why is that? If you wanted to accomplish this, it seems better to have
the URL in the Transfer-To header contain the Call-ID as a URL
parameter. In other words, why not do:

Transfer-To: sip:user@host?Call-ID=9nasd09asd--asdasd98ays


Thanks,
Jonathan R.

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




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From sip-admin@lists.bell-labs.com  Tue May  9 00:19:03 2000
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To: Linden deCarmo <ldeCarmo@netspeak.com>, sip@lists.research.bell-labs.com
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Here's what I would suggest:

Build a SIP UAS that understands the RTSP end and contains stream
management
logic. This UAS either reinvites to establish the link to RTSP streams, or
copies the RTP from each stream back into the media path for the remote UA.

I know of some people(s) who are already using each approach to build a
voice mail system that can use a standard RTSP server as a media store.
Makes unified messaging with URL passing MUCH easier.

--
dean

> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Linden
> deCarmo
> Sent: Tuesday, March 14, 2000 7:27 AM
> To: 'sip@lists.research.bell-labs.com'
> Subject: SIP & RTSP interworking?
>
>
> What would be the proper (or suggested) procedure to tie a SIP
> session to an
> RTSP session?
>
> What I'd like to do is setup a session with SIP (via an INVITE), then
> control streaming aspects via RTSP (i.e. play, pause, stop etc.).
>  However,
> its not clear to me how I tie the RTSP session id to the previously
> established SIP session.
>
> Thanks.
>
> Linden deCarmo
> Netspeak Corporation
> 902 Clint Moore Road
> Suite 104
> Boca Raton, FL 33487
>
>





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Ravandhu Hariram wrote:
>
> After a call is fully connected between two user agents, if one user
agent sends
> a new INVITE request with a different 'Contact' address, should the
receiving
> user agent update its 'Route' header with this modified 'Contact'
address?

I think so. There are a few applications of this, including mobility and
a few call services. Note, however, that updating of the Route with new
Record-Route headers is different. RFC2543 forbids (I think) insertion
of Record-Route headers when a Route is already there.

-Jonathan R.

--
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--NextPart

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

	Title		: MIME media types for ISUP and QSIG Objects
	Author(s)	: E. Zimmerer, A. Vemuri,  L. Ong,
                          M. Zonoun, M. Watson
	Filename	: draft-ietf-sip-isup-mime-01.txt
	Pages		: 6
	Date		: 08-May-00
	
This document describes MIME types for application/ISUP and 
application/QSIG objects for use in SIP applications, according to 
the rules defined in RFC 2048 [1].  These types can be used to identify 
ISUP and QSIG objects within a SIP message such as INVITE or INFO, 
as might be implemented when using SIP between legacy systems.

A URL for this Internet-Draft is:
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Internet-Drafts are also available by anonymous FTP. Login with the username
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--NextPart
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--OtherAccess--

--NextPart--




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From: Vivek Rao <vivek_reo@yahoo.com>
To: sip@lists.research.bell-labs.com
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Hi
   The grammer for contact specified in RFC2543bis is
as follows

Contact   = ("Contact"|"m")":"
                 ("*"|(1#((name_addr|addr_spec)
{*(";"contact-params)])))
name_addr = [display-name]"<"addr_spec">"
addr_spec  = SIP-URL|URI

In the example below 
Contact:<sip:tree@yahoo.com>;action=proxy,gaya<sip:someone@anwhere.com>

Since the grammar for URI field accepts a ';' , '=' &
',' , ";action=proxy,gaya" will be identified as a
URI. Hence its required to scan through this to
retrieve the Contact-Param. All this can be avoided if
we allow a minimum of one space on either side of
comma. 
Similar situation occurs for Via, Route &
Record-Route.

A similar situation occurs if a colon(:) is
immediately followed by a display name as in From or
To fields (e.g. From:Jerry). 

Please clarify on this.
Thanks
Vivek

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To: Vivek Rao <vivek_reo@yahoo.com>
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Subject: Re: [SIP] Sip Grammer: Multiple entries
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--------------AAAEF9D3DC3BF61F60CA2D22
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Content-Transfer-Encoding: 7bit



Vivek Rao wrote:

> Hi
>    The grammer for contact specified in RFC2543bis is
> as follows
>
> Contact   = ("Contact"|"m")":"
>                  ("*"|(1#((name_addr|addr_spec)
> {*(";"contact-params)])))
> name_addr = [display-name]"<"addr_spec">"
> addr_spec  = SIP-URL|URI
>
> In the example below
> Contact:<sip:tree@yahoo.com>;action=proxy,gaya<sip:someone@anwhere.com>
>
> Since the grammar for URI field accepts a ';' , '=' &
> ',' , ";action=proxy,gaya" will be identified as a
> URI. Hence its required to scan through this to
> retrieve the Contact-Param. All this can be avoided if
> we allow a minimum of one space on either side of
> comma.

My interpretation is the URI (or in this case SIP-URL) completes when the
">" is encountered so the "action=proxy" is parsed as a legal
contact-param. So I don't understand the problem.

>
> Similar situation occurs for Via, Route &
> Record-Route.
>
> A similar situation occurs if a colon(:) is
> immediately followed by a display name as in From or
> To fields (e.g. From:Jerry).
>

This one's a bit more problematical because "Jerry" is a valid URI (a
relativeURI), but then again almost anything is. It can't be a display name
because it's not followed by a "<" addr-spec ">". However, I don't think
inserting a space helps, does it? (Inserting white space is always allowed
in headers, but only mandatory when header elements need to be separated,
i.e., between two tokens.)

>
> Please clarify on this.
> Thanks
> Vivek
>
> __________________________________________________
> Do You Yahoo!?
> Send instant messages & get email alerts with Yahoo! Messenger.
> http://im.yahoo.com/
>
> _______________________________________________
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> SIP@lists.bell-labs.com
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Rick Workman
Nortel Networks
Ottawa

--------------AAAEF9D3DC3BF61F60CA2D22
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Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<tt></tt>&nbsp;<tt></tt>
<p><tt>Vivek Rao wrote:</tt>
<blockquote TYPE=CITE><tt>Hi</tt>
<br><tt>&nbsp;&nbsp; The grammer for contact specified in RFC2543bis is</tt>
<br><tt>as follows</tt><tt></tt>
<p><tt>Contact&nbsp;&nbsp; = ("Contact"|"m")":"</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
("*"|(1#((name_addr|addr_spec)</tt>
<br><tt>{*(";"contact-params)])))</tt>
<br><tt>name_addr = [display-name]"&lt;"addr_spec">"</tt>
<br><tt>addr_spec&nbsp; = SIP-URL|URI</tt><tt></tt>
<p><tt>In the example below</tt>
<br><tt>Contact:&lt;sip:tree@yahoo.com>;action=proxy,gaya&lt;sip:someone@anwhere.com></tt><tt></tt>
<p><tt>Since the grammar for URI field accepts a ';' , '=' &amp;</tt>
<br><tt>',' , ";action=proxy,gaya" will be identified as a</tt>
<br><tt>URI. Hence its required to scan through this to</tt>
<br><tt>retrieve the Contact-Param. All this can be avoided if</tt>
<br><tt>we allow a minimum of one space on either side of</tt>
<br><tt>comma.</tt></blockquote>
<tt>My interpretation is the URI (or in this case SIP-URL) completes when
the ">" is encountered so the "action=proxy" is parsed as a legal contact-param.
So I don't understand the problem.</tt>
<blockquote TYPE=CITE><tt></tt>&nbsp;
<br><tt>Similar situation occurs for Via, Route &amp;</tt>
<br><tt>Record-Route.</tt><tt></tt>
<p><tt>A similar situation occurs if a colon(:) is</tt>
<br><tt>immediately followed by a display name as in From or</tt>
<br><tt>To fields (e.g. From:Jerry).</tt>
<br><tt></tt>&nbsp;</blockquote>
<tt>This one's a bit more problematical because "Jerry" is a valid URI
(a relativeURI), but then again almost anything is. It can't be a display
name because it's not followed by a "&lt;" addr-spec ">". However, I don't
think inserting a space helps, does it? (Inserting white space is always
allowed in headers, but only mandatory when header elements need to be
separated, i.e., between two tokens.)</tt>
<blockquote TYPE=CITE><tt></tt>&nbsp;
<br><tt>Please clarify on this.</tt>
<br><tt>Thanks</tt>
<br><tt>Vivek</tt><tt></tt>
<p><tt>__________________________________________________</tt>
<br><tt>Do You Yahoo!?</tt>
<br><tt>Send instant messages &amp; get email alerts with Yahoo! Messenger.</tt>
<br><tt><a href="http://im.yahoo.com/">http://im.yahoo.com/</a></tt><tt></tt>
<p><tt>_______________________________________________</tt>
<br><tt>SIP mailing list</tt>
<br><tt>SIP@lists.bell-labs.com</tt>
<br><tt><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></tt></blockquote>
<tt>Rick Workman</tt>
<br><tt>Nortel Networks</tt>
<br><tt>Ottawa</tt></html>

--------------AAAEF9D3DC3BF61F60CA2D22--



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Reply-To: <albee@miel.mot.com>
From: "Albee Vimal" <albee@miel.mot.com>
To: <sip@lists.research.bell-labs.com>
Date: Tue, 9 May 2000 20:43:48 +0530
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Subject: [SIP] Some clarifications on action field in Contact
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Hi
  We have some doubts on the action parameter that is associated with a
Contact field.
1.We would like to know what it means when we register with action= redirect
or proxy. Does it mean that if a registration is made with action=redirect,
then on receiving an INVITE destined to the registered location, network
server sends a 3XX response to the UAC with all the new contact locations in
the Contact field. The UAC then sends INVITE to all the new locations
mentioned in the Contact field.

2. This is valid only if an UAC can send redirected INVITE to multiple
locations. UAC while forwarding a redirected INVITE replaces the Request-URI
with the  received Contact addresses. If one of the redirected endpoints
responds with a 2XX response, the UAC will be able to forward an ACK to the
Contact field coresponding to it (since the RFC's mandates that a 2XX
response to INVITE contain a Contact field (6.13,  Pg. 39)). But is'nt it
mandatory for an endpoint sending a 6XX response to insert a Contact field
as well.

3. This is related to 2. On receiving a  >3XX response from one of the
endpoints how will the UAC identify the location to which it has to send an
ACK. i.e how would this response be matched with a request transaction,
since the request was sent to multiple destinations.

4. In "16.6 Redirects" example in the RFC we find two Cseq fields in the 302
response. We would like to know whether it is allowed to have multiple CSeq
fields.

Please clarify on these
Thanks
Albee Vimal



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Reply-To: <albee@miel.mot.com>
From: "Albee Vimal" <albee@miel.mot.com>
To: "'Rick Workman'" <rworkman@nortelnetworks.com>
Cc: <sip@lists.research.bell-labs.com>
Subject: RE: [SIP] Sip Grammar: Multiple entries
Date: Tue, 9 May 2000 21:13:42 +0530
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This is a multi-part message in MIME format.

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Thanks for responding to our doubts.
 Since the URI/addr-spec grammar accepts alpha-numeric, ";" ,  "," , "="
etc. We find that during the lexical analysis the rule/token that matches
the maximum number of input is returned  hence ";action=proxy,gaya" is
returned  as a single URI token, instead of being returned as contact-param
,a comma token (comma token refers to ',') and display-name token (refers
the 'gaya'). Earlier we were trying to convey that if a space is introduced
we can avoid this situation and make token retrieval more easy.

Sorry, for giving an incomplete example it should have been
                  From:Jerry<sip:any@doubt>
Here since there is no space between the ':' and 'Jerry',  even though we
specify rule to return From as a separate token and ':' as another. The
lexical analyzer goes for the maximum pattern match and returns "From:Jerry"
as the URI/addr-spec token.  The grammar works fine in identifying
<sip:any@doubt> as URI/addr-spec token. Had there been a space between ':'
and display-name "From:Jerry" would not be returned as a addr-spec token.
Yes, this can be overcome by having some special condition checks and doing
rescan. What we thought was this could be avoided if a space is mandated in
both the cases.
Thanks
Albee Vimal
 -----Original Message-----
From: sip-admin@lists.bell-labs.com [mailto:sip-admin@lists.bell-labs.com]On
Behalf Of Rick Workman
Sent: Tuesday, May 09, 2000 7:47 PM
To: Vivek Rao
Cc: sip@lists.research.bell-labs.com
Subject: Re: [SIP] Sip Grammer: Multiple entries



  Vivek Rao wrote:

    Hi
       The grammer for contact specified in RFC2543bis is
    as follows
    Contact   = ("Contact"|"m")":"
                     ("*"|(1#((name_addr|addr_spec)
    {*(";"contact-params)])))
    name_addr = [display-name]"<"addr_spec">"
    addr_spec  = SIP-URL|URI

    In the example below
    Contact:<sip:tree@yahoo.com>;action=proxy,gaya<sip:someone@anwhere.com>

    Since the grammar for URI field accepts a ';' , '=' &
    ',' , ";action=proxy,gaya" will be identified as a
    URI. Hence its required to scan through this to
    retrieve the Contact-Param. All this can be avoided if
    we allow a minimum of one space on either side of
    comma.

  My interpretation is the URI (or in this case SIP-URL) completes when the
">" is encountered so the "action=proxy" is parsed as a legal contact-param.
So I don't understand the problem.

    Similar situation occurs for Via, Route &
    Record-Route.
    A similar situation occurs if a colon(:) is
    immediately followed by a display name as in From or
    To fields (e.g. From:Jerry).


  This one's a bit more problematical because "Jerry" is a valid URI (a
relativeURI), but then again almost anything is. It can't be a display name
because it's not followed by a "<" addr-spec ">". However, I don't think
inserting a space helps, does it? (Inserting white space is always allowed
in headers, but only mandatory when header elements need to be separated,
i.e., between two tokens.)

    Please clarify on this.
    Thanks
    Vivek
    __________________________________________________
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  Ottawa

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	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">


<META content=3D"MSHTML 5.00.2314.1000" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D868181415-09052000>Thanks=20
for responding to our doubts. </SPAN></FONT></DIV>
<DIV><SPAN class=3D868181415-09052000></SPAN><FONT face=3D"Times New =
Roman"><FONT=20
size=3D2><SPAN class=3D868181415-09052000><FONT color=3D#0000ff =
face=3DArial>&nbsp;Since=20
the URI/addr-spec grammar accepts alpha-numeric, ";" ,&nbsp; "," , "=3D" =
etc. We=20
find that&nbsp;during the lexical analysis the rule/token that matches =
the=20
maximum number of input is returned&nbsp; hence =
";action=3Dproxy,gaya"&nbsp;is=20
returned&nbsp; as a single URI token, instead of being returned as =
contact-param=20
,a comma token (comma token refers to ',') and display-name token =
(refers the=20
'gaya').&nbsp;Earlier we were trying to convey that if a space is =
introduced we=20
can avoid this situation and make token retrieval more=20
easy.</FONT></SPAN></FONT></FONT></DIV>
<DIV><FONT face=3D"Times New Roman"><FONT size=3D2><SPAN=20
class=3D868181415-09052000></SPAN></FONT></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D868181415-09052000>Sorry,=20
for giving an incomplete example it should have been =
</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D868181415-09052000>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
From:Jerry&lt;sip:any@doubt&gt;</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D868181415-09052000>Here=20
since there is no space between the ':' and 'Jerry',  even though we =
specify=20
rule to return From as a separate token and ':' as another. The lexical =
analyzer=20
goes for the maximum pattern match and returns "From:Jerry" as the =
URI/addr-spec=20
token.&nbsp; The grammar works fine in identifying &lt;sip:any@doubt&gt; =
as=20
URI/addr-spec token.&nbsp;Had there been a space between ':' and =
display-name=20
"From:Jerry" would not be returned as a addr-spec token. =
</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D868181415-09052000>Yes,=20
this can be overcome by having some special condition checks and doing =
rescan.=20
What we thought was this could be avoided if a space is mandated in both =
the=20
cases.&nbsp; </SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D868181415-09052000>Thanks</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D868181415-09052000>Albee=20
Vimal</SPAN></FONT></DIV>
<DIV><FONT face=3D"Times New Roman"><FONT size=3D2><SPAN=20
class=3D868181415-09052000>&nbsp;</SPAN>-----Original =
Message-----<BR><B>From:</B>=20
sip-admin@lists.bell-labs.com =
[mailto:sip-admin@lists.bell-labs.com]<B>On Behalf=20
Of</B> Rick Workman<BR><B>Sent:</B> Tuesday, May 09, 2000 7:47 =
PM<BR><B>To:</B>=20
Vivek Rao<BR><B>Cc:</B> =
sip@lists.research.bell-labs.com<BR><B>Subject:</B> Re:=20
[SIP] Sip Grammer: Multiple entries<BR><BR></DIV></FONT>
<BLOCKQUOTE></FONT><TT></TT>&nbsp;<TT></TT>=20
  <P><TT>Vivek Rao wrote:</TT>=20
  <BLOCKQUOTE TYPE=3D"CITE"><TT>Hi</TT> <BR><TT>&nbsp;&nbsp; The grammer =
for=20
    contact specified in RFC2543bis is</TT> <BR><TT>as =
follows</TT><TT></TT>=20
    <P><TT>Contact&nbsp;&nbsp; =3D ("Contact"|"m")":"</TT>=20
    =
<BR><TT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
    ("*"|(1#((name_addr|addr_spec)</TT> =
<BR><TT>{*(";"contact-params)])))</TT>=20
    <BR><TT>name_addr =3D [display-name]"&lt;"addr_spec"&gt;"</TT>=20
    <BR><TT>addr_spec&nbsp; =3D SIP-URL|URI</TT><TT></TT>=20
    <P><TT>In the example below</TT>=20
    =
<BR><TT>Contact:&lt;sip:tree@yahoo.com&gt;;action=3Dproxy,gaya&lt;sip:som=
eone@anwhere.com&gt;</TT><TT></TT>=20

    <P><TT>Since the grammar for URI field accepts a ';' , '=3D' =
&amp;</TT>=20
    <BR><TT>',' , ";action=3Dproxy,gaya" will be identified as a</TT> =
<BR><TT>URI.=20
    Hence its required to scan through this to</TT> <BR><TT>retrieve the =

    Contact-Param. All this can be avoided if</TT> <BR><TT>we allow a =
minimum of=20
    one space on either side of</TT> =
<BR><TT>comma.</TT></P></BLOCKQUOTE><TT>My=20
  interpretation is the URI (or in this case SIP-URL) completes when the =
"&gt;"=20
  is encountered so the "action=3Dproxy" is parsed as a legal =
contact-param. So I=20
  don't understand the problem.</TT>=20
  <BLOCKQUOTE TYPE=3D"CITE"><TT></TT>&nbsp; <BR><TT>Similar situation =
occurs for=20
    Via, Route &amp;</TT> <BR><TT>Record-Route.</TT><TT></TT>=20
    <P><TT>A similar situation occurs if a colon(:) is</TT> =
<BR><TT>immediately=20
    followed by a display name as in From or</TT> <BR><TT>To fields =
(e.g.=20
    From:Jerry).</TT> <BR><TT></TT>&nbsp;</P></BLOCKQUOTE><TT>This one's =
a bit=20
  more problematical because "Jerry" is a valid URI (a relativeURI), but =
then=20
  again almost anything is. It can't be a display name because it's not =
followed=20
  by a "&lt;" addr-spec "&gt;". However, I don't think inserting a space =
helps,=20
  does it? (Inserting white space is always allowed in headers, but only =

  mandatory when header elements need to be separated, i.e., between two =

  tokens.)</TT>=20
  <BLOCKQUOTE TYPE=3D"CITE"><TT></TT>&nbsp; <BR><TT>Please clarify on =
this.</TT>=20
    <BR><TT>Thanks</TT> <BR><TT>Vivek</TT><TT></TT>=20
    <P><TT>__________________________________________________</TT> =
<BR><TT>Do=20
    You Yahoo!?</TT> <BR><TT>Send instant messages &amp; get email =
alerts with=20
    Yahoo! Messenger.</TT> <BR><TT><A=20
    href=3D"http://im.yahoo.com/">http://im.yahoo.com/</A></TT><TT></TT> =

    <P><TT>_______________________________________________</TT> =
<BR><TT>SIP=20
    mailing list</TT> <BR><TT>SIP@lists.bell-labs.com</TT> <BR><TT><A=20
    =
href=3D"http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bel=
l-labs.com/mailman/listinfo/sip</A></TT></P></BLOCKQUOTE><TT>Rick=20
  Workman</TT> <BR><TT>Nortel Networks</TT> <BR><TT>Ottawa</TT>=20
</BLOCKQUOTE></BODY></HTML>

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From: "Lyndon Ong" <long@nortelnetworks.com>
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Folks,

I've sent in updated version of the SIP ISUP MIME type draft
(draft-ietf-sip-isup-mime-01.txt).

Changes as a result of discussion on the sip-t list include the following:

-- ISUP "version" parameter is now mandatory, "base" is added as an
optional parameter
-- some modified wording is given in the security section reflecting
concerns about security of information in the encapsulated messages

A couple of corrections were made based on comments received:

-- some corrections were made to the list of different ISUP base versions
provided in the draft
-- some corrections were made to the ISUP example in section 4.1, esp. a
note that the ISUP message is provided starting with the Message Type Code
(without routing label or CIC).

Cheers,

L. Ong






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Subject: Re: [SIP] Sip Grammar: Multiple entries
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It sounds like you're trying to apply a URI lexical analysis to SIP
headers which isn't going to work too well for the reasons you've
discovered. If you want to take this approach, I think you have to
provide a more granular tokenizer and glue things back together, if
necessary. I suspect most implementations perform context sensitive
parsing, rather than try to do a context insensitive tokenization as a
first step.

Putting in additional spaces will force your tokenizer to work, but
isn't at all backwards compatible with spec or existing implementations.

Rick

Albee Vimal wrote:

>  Thanks for responding to our doubts.  Since the URI/addr-spec grammar
> accepts alpha-numeric, ";" ,  "," , "=" etc. We find that during the
> lexical analysis the rule/token that matches the maximum number of
> input is returned  hence ";action=proxy,gaya" is returned  as a single
> URI token, instead of being returned as contact-param ,a comma token
> (comma token refers to ',') and display-name token (refers the
> 'gaya'). Earlier we were trying to convey that if a space is
> introduced we can avoid this situation and make token retrieval more
> easy.Sorry, for giving an incomplete example it should have
> been                   From:Jerry<sip:any@doubt>Here since there is no
> space between the ':' and 'Jerry', even though we specify rule to
> return From as a separate token and ':' as another. The lexical
> analyzer goes for the maximum pattern match and returns "From:Jerry"
> as the URI/addr-spec token.  The grammar works fine in identifying
> <sip:any@doubt> as URI/addr-spec token. Had there been a space between
> ':' and display-name "From:Jerry" would not be returned as a addr-spec
> token. Yes, this can be overcome by having some special condition
> checks and doing rescan. What we thought was this could be avoided if
> a space is mandated in both the cases. ThanksAlbee Vimal-----Original
> Message-----
> From: sip-admin@lists.bell-labs.com
> [mailto:sip-admin@lists.bell-labs.com]On Behalf Of Rick Workman
> Sent: Tuesday, May 09, 2000 7:47 PM
> To: Vivek Rao
> Cc: sip@lists.research.bell-labs.com
> Subject: Re: [SIP] Sip Grammer: Multiple entries
>
>
>
>
>      Vivek Rao wrote:
>
>     > Hi
>     >    The grammer for contact specified in RFC2543bis is
>     > as follows
>     >
>     > Contact   = ("Contact"|"m")":"
>     >                  ("*"|(1#((name_addr|addr_spec)
>     > {*(";"contact-params)])))
>     > name_addr = [display-name]"<"addr_spec">"
>     > addr_spec  = SIP-URL|URI
>     >
>     > In the example below
>     >
>     > ontact:<sip:tree@yahoo.com>;action=proxy,gaya<sip:someone@anwhere.com>
>     >
>     > Since the grammar for URI field accepts a ';' , '=' &
>     > ',' , ";action=proxy,gaya" will be identified as a
>     > URI. Hence its required to scan through this to
>     > retrieve the Contact-Param. All this can be avoided if
>     > we allow a minimum of one space on either side of
>     > comma.
>
>      My interpretation is the URI (or in this case SIP-URL)
>      completes when the ">" is encountered so the "action=proxy"
>      is parsed as a legal contact-param. So I don't understand
>      the problem.
>
>     >
>     > Similar situation occurs for Via, Route &
>     > Record-Route.
>     >
>     > A similar situation occurs if a colon(:) is
>     > immediately followed by a display name as in From or
>     > To fields (e.g. From:Jerry).
>     >
>
>      This one's a bit more problematical because "Jerry" is a
>      valid URI (a relativeURI), but then again almost anything
>      is. It can't be a display name because it's not followed by
>      a "<" addr-spec ">". However, I don't think inserting a
>      space helps, does it? (Inserting white space is always
>      allowed in headers, but only mandatory when header elements
>      need to be separated, i.e., between two tokens.)
>
>     >
>     > Please clarify on this.
>     > Thanks
>     > Vivek
>     >
>     > __________________________________________________
>     > Do You Yahoo!?
>     > Send instant messages & get email alerts with Yahoo!
>     > Messenger.
>     > http://im.yahoo.com/
>     >
>     > _______________________________________________
>     > SIP mailing list
>     > SIP@lists.bell-labs.com
>     > http://lists.bell-labs.com/mailman/listinfo/sip
>
>      Rick Workman
>      Nortel Networks
>      Ottawa
>

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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
&nbsp;
<br>It sounds like you're trying to apply a URI lexical analysis to SIP
headers which isn't going to work too well for the reasons you've discovered.
If you want to take this approach, I think you have to provide a more granular
tokenizer and glue things back together, if necessary. I suspect most implementations
perform context sensitive parsing, rather than try to do a context insensitive
tokenization as a first step.
<p>Putting in additional spaces will force your tokenizer to work, but
isn't at all backwards compatible with spec or existing implementations.
<p>Rick
<p>Albee Vimal wrote:
<blockquote TYPE=CITE>&nbsp;<span class=868181415-09052000><font face="Arial"><font color="#0000FF"><font size=-1>Thanks
for responding to our doubts.&nbsp;</font></font></font></span><span class=868181415-09052000></span><span class=868181415-09052000><font face="Arial"><font color="#0000FF"><font size=-1>
Since the URI/addr-spec grammar accepts alpha-numeric, ";" ,&nbsp; ","
, "=" etc. We find that during the lexical analysis the rule/token that
matches the maximum number of input is returned&nbsp; hence ";action=proxy,gaya"
is returned&nbsp; as a single URI token, instead of being returned as contact-param
,a comma token (comma token refers to ',') and display-name token (refers
the 'gaya'). Earlier we were trying to convey that if a space is introduced
we can avoid this situation and make token retrieval more easy.</font></font></font></span><span 
class=868181415-09052000></span><span class=868181415-09052000><font face="Arial"><font color="#0000FF"><font size=-1>Sorry,
for giving an incomplete example it should have been&nbsp;</font></font></font></span><span 
class=868181415-09052000><font face="Arial"><font color="#0000FF"><font size=-1>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
From:Jerry&lt;sip:any@doubt></font></font></font></span><span class=868181415-09052000><font face="Arial"><font color="#0000FF"><font size=-1>Here
since there is no space between the ':' and 'Jerry', even though we specify
rule to return From as a separate token and ':' as another. The lexical
analyzer goes for the maximum pattern match and returns "From:Jerry" as
the URI/addr-spec token.&nbsp; The grammar works fine in identifying &lt;sip:any@doubt>
as URI/addr-spec token. Had there been a space between ':' and display-name
"From:Jerry" would not be returned as a addr-spec token.&nbsp;</font></font></font></span><span class=868181415-09052000><font face="Arial"><font color="#0000FF"><font size=-1>Yes,
this can be overcome by having some special condition checks and doing
rescan. What we thought was this could be avoided if a space is mandated
in both the cases.&nbsp;</font></font></font></span><span 
class=868181415-09052000><font face="Arial"><font color="#0000FF"><font size=-1>Thanks</font></font></font></span><span class=868181415-09052000><font face="Arial"><font color="#0000FF"><font size=-1>Albee
Vimal</font></font></font></span><span 
class=868181415-09052000></span><font face="Times New Roman"><font size=-1>-----Original
Message-----</font></font>
<br><font face="Times New Roman"><font size=-1><b>From:</b> sip-admin@lists.bell-labs.com
[<A HREF="mailto:sip-admin@lists.bell-labs.com">mailto:sip-admin@lists.bell-labs.com</A>]<b>On Behalf Of</b> Rick Workman</font></font>
<br><font face="Times New Roman"><font size=-1><b>Sent:</b> Tuesday, May
09, 2000 7:47 PM</font></font>
<br><font face="Times New Roman"><font size=-1><b>To:</b> Vivek Rao</font></font>
<br><font face="Times New Roman"><font size=-1><b>Cc:</b> sip@lists.research.bell-labs.com</font></font>
<br><font face="Times New Roman"><font size=-1><b>Subject:</b> Re: [SIP]
Sip Grammer: Multiple entries</font></font>
<br>&nbsp;
<blockquote>&nbsp;
<p><tt>Vivek Rao wrote:</tt>
<blockquote TYPE="CITE"><tt>Hi</tt>
<br><tt>&nbsp;&nbsp; The grammer for contact specified in RFC2543bis is</tt>
<br><tt>as follows</tt>
<p><tt>Contact&nbsp;&nbsp; = ("Contact"|"m")":"</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
("*"|(1#((name_addr|addr_spec)</tt>
<br><tt>{*(";"contact-params)])))</tt>
<br><tt>name_addr = [display-name]"&lt;"addr_spec">"</tt>
<br><tt>addr_spec&nbsp; = SIP-URL|URI</tt>
<p><tt>In the example below</tt>
<br><tt>Contact:&lt;sip:tree@yahoo.com>;action=proxy,gaya&lt;sip:someone@anwhere.com></tt>
<p><tt>Since the grammar for URI field accepts a ';' , '=' &amp;</tt>
<br><tt>',' , ";action=proxy,gaya" will be identified as a</tt>
<br><tt>URI. Hence its required to scan through this to</tt>
<br><tt>retrieve the Contact-Param. All this can be avoided if</tt>
<br><tt>we allow a minimum of one space on either side of</tt>
<br><tt>comma.</tt></blockquote>
<tt>My interpretation is the URI (or in this case SIP-URL) completes when
the ">" is encountered so the "action=proxy" is parsed as a legal contact-param.
So I don't understand the problem.</tt>
<blockquote TYPE="CITE">&nbsp;
<br><tt>Similar situation occurs for Via, Route &amp;</tt>
<br><tt>Record-Route.</tt>
<p><tt>A similar situation occurs if a colon(:) is</tt>
<br><tt>immediately followed by a display name as in From or</tt>
<br><tt>To fields (e.g. From:Jerry).</tt>
<br>&nbsp;</blockquote>
<tt>This one's a bit more problematical because "Jerry" is a valid URI
(a relativeURI), but then again almost anything is. It can't be a display
name because it's not followed by a "&lt;" addr-spec ">". However, I don't
think inserting a space helps, does it? (Inserting white space is always
allowed in headers, but only mandatory when header elements need to be
separated, i.e., between two tokens.)</tt>
<blockquote TYPE="CITE">&nbsp;
<br><tt>Please clarify on this.</tt>
<br><tt>Thanks</tt>
<br><tt>Vivek</tt>
<p><tt>__________________________________________________</tt>
<br><tt>Do You Yahoo!?</tt>
<br><tt>Send instant messages &amp; get email alerts with Yahoo! Messenger.</tt>
<br><tt><a href="http://im.yahoo.com/">http://im.yahoo.com/</a></tt>
<p><tt>_______________________________________________</tt>
<br><tt>SIP mailing list</tt>
<br><tt>SIP@lists.bell-labs.com</tt>
<br><tt><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></tt></blockquote>
<tt>Rick Workman</tt>
<br><tt>Nortel Networks</tt>
<br><tt>Ottawa</tt></blockquote>
</blockquote>
</html>

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From: Jonathan Lennox <lennox@cs.columbia.edu>
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Subject: RE: [SIP] Sip Grammar: Multiple entries
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On Tue, May 9 2000, "Albee Vimal" wrote to "'Rick Workman', sip@lists.research.bell-labs.com" saying:

> Thanks for responding to our doubts.
>  Since the URI/addr-spec grammar accepts alpha-numeric, ";" ,  "," , "="
> etc. We find that during the lexical analysis the rule/token that matches
> the maximum number of input is returned  hence ";action=proxy,gaya" is
> returned  as a single URI token, instead of being returned as contact-param
> ,a comma token (comma token refers to ',') and display-name token (refers
> the 'gaya'). Earlier we were trying to convey that if a space is introduced
> we can avoid this situation and make token retrieval more easy.

The simplest solution to this is to drop ',' from the set of characters
which is allowed to appear unescaped in a SIP URI, as Igor suggested a few
days ago.  I don't think this will cause much in the way of compatibility
problems (though you SHOULD handle an unescaped comma in a URI correctly if
it's between angle brackets, in a Request-URI, or whatever).


> Sorry, for giving an incomplete example it should have been
>                   From:Jerry<sip:any@doubt>
> Here since there is no space between the ':' and 'Jerry',  even though we
> specify rule to return From as a separate token and ':' as another. The
> lexical analyzer goes for the maximum pattern match and returns "From:Jerry"
> as the URI/addr-spec token.  The grammar works fine in identifying
> <sip:any@doubt> as URI/addr-spec token. Had there been a space between ':'
> and display-name "From:Jerry" would not be returned as a addr-spec token.
> Yes, this can be overcome by having some special condition checks and doing
> rescan. What we thought was this could be avoided if a space is mandated in
> both the cases.

This one, however, I don't think is a problem.  Tracing down the grammar for
the generic message-header syntax in 2543, we find:

        message-header  =  field-name ":" [ field-value ] CRLF
        field-name      =  token

        token       =  1*< any CHAR  except CTL's  or separators>
        separators  =  "(" | ")" | "<" | ">" | "@" |
                       "," | ";" | ":" | "\" | <"> |
                       "/" | "[" | "]" | "?" | "=" |
                       "{" | "}" | SP | HT

Thus, ":" isn't allowed in a field-name, so the grammar is unambiguous.

It looks like what you're trying to do is to have a separate tokenizer and
parser (the usual lex/yacc split)?   I don't think the SIP grammar will
support that, unfortunately; it's too irregular at a low level.

-- 
Jonathan Lennox
lennox@cs.columbia.edu


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Date: Tue, 09 May 2000 12:59:31 -0700
From: Marc Petit-Huguenin <petithug@8x8.com>
Organization: Netergy Networks - Advanced Telephony Solutions
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Subject: [SIP] draft 2543Bis
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Hi,

Some comments on 2543Bis:

In the SIP URL syntax (figure 3) the new user parameter "np-queried" is
not defined.

Also in the Client error status codes (figure 7), the 487 code is not
defined.

And somes questions:

The second paragraph of the section 4.2.4 states that "the INVITE must
be completed with a final response". But which response do we must use?
A 487 like for a CANCEL or another response?

The first paragraph of the same section does not explain when the sender
of the BYE request must cease to transmit media streams. Is it when the
BYE is sent or when the 200 is received?

Thank you for your responses,

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


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From: "Joshua Moloney" <jmoloney@nortelnetworks.com>
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Hi all,

Having skipped through the new release of the draft specifying the rules
surrounding ISUP & QSIG MIME types, I have a question...

It is my understanding that the Content-Transfer-Encoding header was
primarily used to indicate that a message body had been transformed into a
format that wasn't 8-bit clean (thanks to Jonathan R. for explaining that!).
SIP is 8-bit clean (so I'm told!), so here's the rub - why is the
Content-Transfer-Encoding header used to specify binary, when we know it's
going to be binary anyway? Can't we ditch it?

Cheers,

Josh Moloney
Nortel Networks


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<P><FONT SIZE=3D2 FACE=3D"Arial">Hi all,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Having skipped through the new release =
of the draft specifying the rules surrounding ISUP &amp; QSIG MIME =
types, I have a question...</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">It is my understanding that the =
Content-Transfer-Encoding header was primarily used to indicate that a =
message body had been transformed into a format that wasn't 8-bit clean =
(thanks to Jonathan R. for explaining that!). SIP is 8-bit clean (so =
I'm told!), so here's the rub - why is the Content-Transfer-Encoding =
header used to specify binary, when we know it's going to be binary =
anyway? Can't we ditch it?</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Cheers,</FONT>
</P>

<P><FONT COLOR=3D"#000000" SIZE=3D2 FACE=3D"Arial">Josh Moloney</FONT>
<BR><FONT COLOR=3D"#000000" SIZE=3D2 FACE=3D"Arial">Nortel =
Networks</FONT>
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Subject: Implementing H.323 
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Implementing H.323 : conference=20
10-13 October 2000
CFP online at:
http://www.upperside.fr/bah323.htm

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From sip-admin@lists.bell-labs.com  Wed May 10 07:13:48 2000
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Hi,

The BIS draft modified paragraph 10.2.2 Multicast UDP, to include
the 401 status code in the following sentence ....

"To avoid response implosion, servers MUST NOT answer multicast
requests with a status code other than 2xx, 401 or 6xx."

Should the last sentence of the same paragraph also be modified
to read .....

"A proxy or UAC SHOULD send a CANCEL on receiving the first 2xx,
401 or 6xx response to a multicast request"
---
 -

Without the addition of the 401 to the above sentence, the UAC
processing of multiple responses from a multicast request becomes
confused, as a 401 can arrive first.

Thanks In Advance
Phil

--
Phil Hoffer                         Ubiquity Software Corporation
Software Developer                  Ubiquity House
Tel: +44 (0) 1633-765600            Langstone Park
Fax: +44 (0) 1633-765601            Newport   NP18 2LH
URL: http://www.ubiquity.net




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From sip-admin@lists.bell-labs.com  Wed May 10 07:52:53 2000
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Yes; changed.

Phil Hoffer wrote:
> 
> Hi,
> 
> The BIS draft modified paragraph 10.2.2 Multicast UDP, to include
> the 401 status code in the following sentence ....
> 
> "To avoid response implosion, servers MUST NOT answer multicast
> requests with a status code other than 2xx, 401 or 6xx."
> 
> Should the last sentence of the same paragraph also be modified
> to read .....
> 
> "A proxy or UAC SHOULD send a CANCEL on receiving the first 2xx,
> 401 or 6xx response to a multicast request"
> ---
>  -
> 
> Without the addition of the 401 to the above sentence, the UAC
> processing of multiple responses from a multicast request becomes
> confused, as a 401 can arrive first.
>


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From sip-admin@lists.bell-labs.com  Wed May 10 08:05:10 2000
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Hi Henning,

Thanks for the quick reply.

Following on from this .....

I believe a decision was made post bake off to allow a registrar to
return a 407, in response to a register request.

This of course has implications with respect to the handling of multicast
registers, as the paragraphs quoted only refer to 401.

I think the options are to:

a) Reverse the post bake off decision and state that a registrar SHOULD
ONLY return 401 if it requires authentication.

b) Modify the quoted paragraphs to also include a 407 response.

What do you think?

Thanks In Advance
Phil

Henning Schulzrinne wrote:

> Yes; changed.
>
> Phil Hoffer wrote:
> >
> > Hi,
> >
> > The BIS draft modified paragraph 10.2.2 Multicast UDP, to include
> > the 401 status code in the following sentence ....
> >
> > "To avoid response implosion, servers MUST NOT answer multicast
> > requests with a status code other than 2xx, 401 or 6xx."
> >
> > Should the last sentence of the same paragraph also be modified
> > to read .....
> >
> > "A proxy or UAC SHOULD send a CANCEL on receiving the first 2xx,
> > 401 or 6xx response to a multicast request"
> > ---
> >  -
> >
> > Without the addition of the 401 to the above sentence, the UAC
> > processing of multiple responses from a multicast request becomes
> > confused, as a 401 can arrive first.
> >

--
Phil Hoffer                         Ubiquity Software Corporation
Software Developer                  Ubiquity House
Tel: +44 (0) 1633-765600            Langstone Park
Fax: +44 (0) 1633-765601            Newport   NP18 2LH
URL: http://www.ubiquity.net




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From sip-admin@lists.bell-labs.com  Wed May 10 09:40:30 2000
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Subject: Re: [SIP] Multicast UDP Authentication
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Hi again,

Is it possible for a proxy server to proxy a multicast request?

If so then that would suggest that Proxy-Authenticate (407) might
be required in order to achieve this.

Therefore option b) would seem reasonable.

Thoughts??

Regards
Phil

Phil Hoofer wrote:

> Hi Henning,
>
> Thanks for the quick reply.
>
> Following on from this .....
>
> I believe a decision was made post bake off to allow a registrar to
> return a 407, in response to a register request.
>
> This of course has implications with respect to the handling of multicast
> registers, as the paragraphs quoted only refer to 401.
>
> I think the options are to:
>
> a) Reverse the post bake off decision and state that a registrar SHOULD
> ONLY return 401 if it requires authentication.
>
> b) Modify the quoted paragraphs to also include a 407 response.
>
> What do you think?
>
> Thanks In Advance
> Phil
>
> Henning Schulzrinne wrote:
>
> > Yes; changed.
> >
> > Phil Hoffer wrote:
> > >
> > > Hi,
> > >
> > > The BIS draft modified paragraph 10.2.2 Multicast UDP, to include
> > > the 401 status code in the following sentence ....
> > >
> > > "To avoid response implosion, servers MUST NOT answer multicast
> > > requests with a status code other than 2xx, 401 or 6xx."
> > >
> > > Should the last sentence of the same paragraph also be modified
> > > to read .....
> > >
> > > "A proxy or UAC SHOULD send a CANCEL on receiving the first 2xx,
> > > 401 or 6xx response to a multicast request"
> > > ---
> > >  -
> > >
> > > Without the addition of the 401 to the above sentence, the UAC
> > > processing of multiple responses from a multicast request becomes
> > > confused, as a 401 can arrive first.
> > >

--
Phil Hoffer                         Ubiquity Software Corporation
Software Developer                  Ubiquity House
Tel: +44 (0) 1633-765600            Langstone Park
Fax: +44 (0) 1633-765601            Newport   NP18 2LH
URL: http://www.ubiquity.net




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Phil Hoffer wrote:
> 
> Hi Henning,
> 
> Thanks for the quick reply.
> 
> Following on from this .....
> 
> I believe a decision was made post bake off to allow a registrar to
> return a 407, in response to a register request.

No.

A registrar *terminates* the request. It does not proxy it. Therefore,
it uses 401. A proxy which sends a REGISTER to a registrar is *proxying*
the request, and thus uses 407, NOT 401. That was the clarification made
during the bakeoff.

> b) Modify the quoted paragraphs to also include a 407 response.

Actually, the quoted paragraphs need to be modified to handle any
response code for which a subsequent request is needed and requested.
This includes 401, 407, and 484.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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In an attempt to get some closure on the SIP-URL header BNF, how about:

headers     = "?" header *("&" header)
header      = hname "=" hvalue
hname       = 1*(hname_char)
hvalue      = *(hvalue_char)

hname_char  = alphanum | "-" | "." | "!" | "*" | "_" | "+" | "'" | "~"

hvalue_char = hvreserved | unreserved | escaped
hvreserved  = ";" | "/" | "?" | ":" | "@" | "=" | "+" |"$"


Notes:
1. This isn't strictly backward compatible but I can't see any practical
issues. (Same as proposed change to other-param?)
2. hname_char set is a subset of token (to be consistent with
"field-name") which doesn't include "%" (URI "delims") or "`" (URI
"unwise").
3. hvreserved is a subset of the current reserved rule eliminating "&",
"," as per earlier discussion, and "[" and "]" (both URI "unwise").
4. Scanning the current header definitions, at least the following must
be escaped:
   - any required LWS, e.g., for token separation or in rfc1123 dates
   - double quotes in quoted strings
   - "\" in quoted-pair (comments and quoted-strings)
   - "%" and "`" in tokens
   - "<" and ">" in name-addrs
   - "&" in any embedded SIP-URL headers
   - "," in # lists
   - any illegal characters in TEXT-UTF8
   - any others?
5. Multi-byte UTF-8 characters will be represent by a 2*escaped , i.e.,
the UTF-8 character CA98 will escape to %CA%98. (I think this was the
intent of Jonathan Lennox's previous proposal.)


Also, I'd like to see the text concerning reserved characters and
escaping clarified. If the BNF can be tightened up to the point where the
allowable characters for any SIP-URL component have been clearly defined,
then, by definition, any other characters must be escaped. Calling these
other characters reserved confuses them with those in the "reserved" rule
in the URI spec (and the current SIP RFC). Am I missing something?


Rick Workman
Nortel Networks
Ottawa

--------------9F7BF062EF6473DA11D8978A
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
&nbsp;
<br><tt>In an attempt to get some closure on the SIP-URL header BNF, how
about:</tt>
<p><tt>headers&nbsp;&nbsp;&nbsp;&nbsp; = "?" header *("&amp;" header)</tt>
<br><tt>header&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = hname "=" hvalue</tt>
<br><tt>hname&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = 1*(hname_char)</tt>
<br><tt>hvalue&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = *(hvalue_char)</tt><tt></tt>
<p><tt>hname_char&nbsp; = alphanum | "-" | "." | "!" | "*" | "_" | "+"
| "'" | "~"</tt>
<p><tt>hvalue_char = hvreserved | unreserved | escaped</tt>
<br><tt>hvreserved&nbsp; = ";" | "/" | "?" | ":" | "@" | "=" | "+" |"$"</tt>
<br><tt></tt>&nbsp;<tt></tt>
<p><tt>Notes:</tt>
<br><tt>1. This isn't strictly backward compatible but I can't see any
practical issues. (Same as proposed change to other-param?)</tt>
<br><tt>2. hname_char set is a subset of token (to be consistent with "field-name")
which doesn't include "%" (URI "delims") or "`" (URI "unwise").</tt>
<br><tt>3. hvreserved is a subset of the current reserved rule eliminating
"&amp;", "," as per earlier discussion, and "[" and "]" (both URI "unwise").</tt>
<br><tt>4. Scanning the current header definitions, at least the following
must be escaped:</tt>
<br><tt>&nbsp;&nbsp; - any required LWS, e.g., for token separation or
in rfc1123 dates</tt>
<br><tt>&nbsp;&nbsp; - double quotes in quoted strings</tt>
<br><tt>&nbsp;&nbsp; - "\" in quoted-pair (comments and quoted-strings)</tt>
<br><tt>&nbsp;&nbsp; - "%" and "`" in tokens</tt>
<br><tt>&nbsp;&nbsp; - "&lt;" and ">" in name-addrs</tt>
<br><tt>&nbsp;&nbsp; - "&amp;" in any embedded SIP-URL headers</tt>
<br><tt>&nbsp;&nbsp; - "," in # lists</tt>
<br><tt>&nbsp;&nbsp; - any illegal characters in TEXT-UTF8</tt>
<br><tt>&nbsp;&nbsp; - any others?</tt>
<br><tt>5. Multi-byte UTF-8 characters will be represent by a 2*escaped
, i.e., the UTF-8 character CA98 will escape to %CA%98. (I think this was
the intent of Jonathan Lennox's previous proposal.)</tt>
<br><tt></tt>&nbsp;<tt></tt>
<p><tt>Also, I'd like to see the text concerning reserved characters and
escaping clarified. If the BNF can be tightened up to the point where the
allowable characters for any SIP-URL component have been clearly defined,
then, by definition, any other characters must be escaped. Calling these
other characters reserved confuses them with those in the "reserved" rule
in the URI spec (and the current SIP RFC). Am I missing something?</tt>
<br><tt></tt>&nbsp;<tt></tt>
<p><tt>Rick Workman</tt>
<br><tt>Nortel Networks</tt>
<br><tt>Ottawa</tt></html>

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Phil Hoffer wrote:
> 
> > > 2) A transition from Failure state to Initial state is shown on the diagram, as
> > > a dashed line. Unfortunately, this transition is unlabelled. I'm referring
> > > to the left hand dashed line, not the BYE event which joins the Success
> > > dashed line.
> >
> > Hmm. not sure, actually. The labeling would make this appear to be
> > CANCEL, but thats not correct. Henning?
> 
> Presumably this unlabelled event could be a CANCEL as they UAC
> might have issued the CANCEL when the UAS was in Call Proceeding
> state. So the >= 300 response could cross the incoming CANCEL on
> the wire. Therefore a CANCEL could arrive in the Failure state. ??!!

Yes, but in this case the next state is not Initial. You've already sent
the response, and need to retransmit until the ACK comes.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
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Jonathan Lennox wrote:
> 
> On Tue, May 9 2000, "Albee Vimal" wrote to "'Rick Workman', sip@lists.research.bell-labs.com" saying:
> 
> > Thanks for responding to our doubts.
> >  Since the URI/addr-spec grammar accepts alpha-numeric, ";" ,  "," , "="
> > etc. We find that during the lexical analysis the rule/token that matches
> > the maximum number of input is returned  hence ";action=proxy,gaya" is
> > returned  as a single URI token, instead of being returned as contact-param
> > ,a comma token (comma token refers to ',') and display-name token (refers
> > the 'gaya'). Earlier we were trying to convey that if a space is introduced
> > we can avoid this situation and make token retrieval more easy.
> 
> The simplest solution to this is to drop ',' from the set of characters
> which is allowed to appear unescaped in a SIP URI, as Igor suggested a few
> days ago.  I don't think this will cause much in the way of compatibility
> problems (though you SHOULD handle an unescaped comma in a URI correctly if
> it's between angle brackets, in a Request-URI, or whatever).

I still don't get how your analyzer figured that ;action=proxy,gaya is a
URI. Was sip:tree@yahoo.com picked up as a URI? If so, the grammar
clearly requires a comma between these two.

In any case, I really don't want to mandate white space between tokens
and separators, particularly in random select places like this. The rule
for white space is generic and applies everywhere. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
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Rick Workman wrote:
> 
> 
> 
> Sean Olson wrote:
> 
> > > The current SIP BNF allows the general form of URI (or
> > URI-reference?) to
> > > be used in the request line and various headers. Observations and
> > > questions:
> > >
> > > 1. A valid form of URI (RFC 2396)is the empty sequence since:
> > >
> > > URI-reference = [ absoluteURI | relativeURI ] [ "#" fragment ]
> > >
> > > This would permit the request line and various headers (To, From,
> > > Contact. ..) without an address specification which doesn't seem
> > to make
> > > much sense. RFC 2396 specifies that the empty sequence URI refers
> > to the
> > > start of the current document; is there any corresponding
> > abstraction for
> > > SIP transcations?
> >
> > No. Please god no. Obviously RFC2396 specifies a much wider and more
> > abstract form
> > of URI than is commonly used in SIP. Not everything in 2396 is
> > appropriate for
> > every URI scheme. They are a set of guidelines that work more or
> > less for the most
> > common URI schemes in use today: http, ftp, telnet, mailto (with
> > some quirks), etc.
> >
> 
> Be that as it may, they're guidelines which are now an integral part
> of the SIP BNF.

Actually, this doesn't seem to be the case. The BNF for request URI, for
example, is never defined in RFC2543! It says in section 4.3 thats its
either a SIP URI or general URI. The BNF for addr-spec (used in To,
From, Contact) is defined as SIP-URL or URI, but I also never found the
BNF for URI. Even rfc2396 doesn't define "URI", but rather absoluteURI,
relativeURI or URI-reference. 

I believe what URI really means in the SIP context is absoluteURI. That
is, there must always be a scheme, followed by a colon. I believe this
eliminates many of the ambiguities people have been raising. I'm not a
big fan of relative URIs within HTTP itself (as opposed to within HTML,
where the browser computes the absolute URI and puts that in the HTTP
request); in general, I believe URIs sent on the wire should be globally
resolvable. HTTP ran into all sorts of problems, as I recall, since URIs
were not always absolute, and thus servers that supported many domains
got confused when they received things like GET index.html HTTP/1.0.

> 
> >
> >
> > >
> > > 2. The relative URI references a resource relative to a base URI
> > of a
> > > hierarchial scheme. So a similar question: is this a useful
> > concept for
> > > SIP sessions?
> > >
> >
> > The only contexts where I would think this might be useful would be
> > in
> > 1) The headers parameter of a SIP URI (for the From: and possibly
> > To: headers)
> > 2) A SIP "cookie" similar to an HTTP-cookie. This hasn't really been
> > investigated
> >    much but would be interesting.
> >
> > In other words, I don't think it is generally useful in the usual
> > places in a SIP
> > message (To/From/Contact/Request-URI). But it is potentially useful
> > at the user interface
> > level or as part of some service logic.
> 
> As part of a SIP message? I guess I can think of many examples where
> you want to map user friendly "addresses" to URI's in SIP messages,
> but that's out of scope for this discussion.

I really don't see the use of relative URIs within SIP itself.

> 
> >
> >
> > > 3. Finally RFC 2396 states (my emphasis):
> > >   "When a URI reference is used to perform a retrieval action on
> > the
> > >    identified resource, the optional fragment identifier,
> > separated from
> > >    the URI by a crosshatch ("#") character, consists of additional
> >
> > >    reference information to be interpreted by the user agent after
> > the
> > >    retrieval action has been successfully completed.  As such, it
> > is not
> > >    part of a URI, but is often used in conjunction with a URI."
> > >
> >
> > True, but this treated in very different ways by different web
> > browsers. Some
> > treat it as part of the URI, others use it for "post-processing".
> > It's not really used in a standard way for SIP, but as stated above
> > it really is up
> > to the UserAgent's interpretation. THIS IS A GOOD THING.
> 
> I'm not arguing the merits of the URI syntax, just how it's used in
> SIP messages.

Again, if we say that URI = absoluteURI, this problem goes away also.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Albee Vimal wrote:
> 
> Hi
>   We have some doubts on the action parameter that is associated with a
> Contact field.
> 1.We would like to know what it means when we register with action= redirect
> or proxy. Does it mean that if a registration is made with action=redirect,
> then on receiving an INVITE destined to the registered location, network
> server sends a 3XX response to the UAC with all the new contact locations in
> the Contact field. The UAC then sends INVITE to all the new locations
> mentioned in the Contact field.

It doesn't have to send the INVITE to all the new locations, but besides
that, yes.

> 
> 2. This is valid only if an UAC can send redirected INVITE to multiple
> locations. UAC while forwarding a redirected INVITE replaces the Request-URI
> with the  received Contact addresses. If one of the redirected endpoints
> responds with a 2XX response, the UAC will be able to forward an ACK to the
> Contact field coresponding to it (since the RFC's mandates that a 2XX
> response to INVITE contain a Contact field (6.13,  Pg. 39)). But is'nt it
> mandatory for an endpoint sending a 6XX response to insert a Contact field
> as well.

The rules here for sending ACK are the same for any other case. You
build the Route header as you normally would (consisting of the Contact
and/or Record-Route headers), and use that. If there wasn't a Contact or
Record-Route, the ACK goes to the same place the INVITE went to.


> 
> 3. This is related to 2. On receiving a  >3XX response from one of the
> endpoints how will the UAC identify the location to which it has to send an
> ACK. i.e how would this response be matched with a request transaction,
> since the request was sent to multiple destinations.

Branch id in the via header, just like a forking proxy.

> 
> 4. In "16.6 Redirects" example in the RFC we find two Cseq fields in the 302
> response. We would like to know whether it is allowed to have multiple CSeq
> fields.

Oops. This is wrong. Only one CSeq is allowed.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Manoj Bhatia wrote:
> 
> Hi Jonathan
> 
> Can we at least draw some consensus on the 'Meesage Waiting
> Indication(MWI)'
> event in Voice mail context ?

I think there is a clear need for subscription and notification events
for many things, including message waiting indication. The question is
how to structure this in the most general way. I really don't want a
specific and separate spec just for MWI, as it seems to narrow.

> 
> Further, in one of the threads on this list, Henning says that we must
> adopt
> drafts/discussion on PINT to decide on MWI.

To me, what this means is that:

1. we use the SUBSCRIBE and NOTIFY methods
2. the body is used to define the object being subscribed to


> 
> Also, there is some concern that whether SUBSCRIBE -  a new SIP method
> 
> should be used for sending subscription requests or is REGISTER a good
> way of doing that ?

Definitely not REGISTER. Its semantics are completely different. It
updates a binding, in a server, of the address in the To with the
addresses in the Contact. I don't see at all how this accomplishes MWI.

> 
> My concern is that very soon you may have interop problems with
> SUBSCRIBE/
> REGISTER method being used by vendors in different ways .

You should absolutely not be using REGISTER for this.

> 
> Is SUBSCRIBE/NOTIFY the de facto way for MWI implementation ?

I think it is clearly the basis. The question, again, is how to do this
generally.

Some of the questions in my mind are:

1. how do we handle addressing? I.e., what is the entity that is being
subscribed to, in the request URI, for mwi? Is it even a SIP URL?

2. What is the format of the content in the body for mwi? Is there a
format which is more generic than just mwi?

3. What is the format of the content in notifications?

4. security issues - how do you prevent attacks whereby I subscribe to
some events, and list someone else in the Contact, say, to receive the
notifications?

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Phil Hoffer wrote:
> 
> Hi again,
> 
> Is it possible for a proxy server to proxy a multicast request?

Of course.

> 
> If so then that would suggest that Proxy-Authenticate (407) might
> be required in order to achieve this.
> 
> Therefore option b) would seem reasonable.

Yes.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Marc Petit-Huguenin wrote:
> 
> Hi,
> 
> Some comments on 2543Bis:
> 
> In the SIP URL syntax (figure 3) the new user parameter "np-queried" is
> not defined.

Was there consensus this was the right way to do this? I recall we were
looking for something maybe more general, but I don't think the
discussion converged.

> And somes questions:
> 
> The second paragraph of the section 4.2.4 states that "the INVITE must
> be completed with a final response". But which response do we must use?
> A 487 like for a CANCEL or another response?

Doesn't actually matter. 4xx is probably fine.

> 
> The first paragraph of the same section does not explain when the sender
> of the BYE request must cease to transmit media streams. Is it when the
> BYE is sent or when the 200 is received?

Practically speaking, if a UA sends a BYE, the user wants to end the
call, so you should probably stop sending audio right away. However, the
SIP machinery still needs to hang on for a bit. Its possible that a BYE
can generate a 5xx response with a retry-after; this can happen in a
particular condition when the UAS answers the phone and hangs up
immediately, and then the 200 OK and BYE pass on the wire. The caller
gets the BYE before getting the 200, and can't match it to a call leg
since it hasn't gotten the tag in the To yet to its original INVITE. So,
the right thing to do is respond to the BYE with retry-after to give the
200 OK a chance to arrive.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Yes, I believe it should be removed.

-Jonathan R.

> Joshua Moloney wrote:
> 
> Hi all,
> 
> Having skipped through the new release of the draft specifying the
> rules surrounding ISUP & QSIG MIME types, I have a question...
> 
> It is my understanding that the Content-Transfer-Encoding header was
> primarily used to indicate that a message body had been transformed
> into a format that wasn't 8-bit clean (thanks to Jonathan R. for
> explaining that!). SIP is 8-bit clean (so I'm told!), so here's the
> rub - why is the Content-Transfer-Encoding header used to specify
> binary, when we know it's going to be binary anyway? Can't we ditch
> it?
> 
> Cheers,
> 
> Josh Moloney
> Nortel Networks

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Folks,

Enclosed are draft minutes of the SIp security task force meeting in
Adelaide. Remember, "consensus" here does not represent consensus of the
working group and this should be looked at more as a first step
discussion.

Thanks to Flemming for taking and preparing these minutes.

-Jonathan R.



Minutes of SIP Security Task Force meeting 
========================================== 
47th IETF meeting, Adelaide 

Prepared by Flemming Andreasen 

A number of people participated in the SIP Security Task Force meeting
in Adelaide on
Sunday 3/25/00 before the IETF meeting started. The purpose of the
meeting was to start
discussion on SIP security. 

1. Goal 
======= 
The following goals were established for the meeting: 
- Identify threats, what are the problems we are trying to solve: 
  o Authorization of callers, user-user authentication model 
  o SIP for secure media streams, encryption of media streams 

The following goal was suggested as well, however it was decided it may
not be in scope
for now: 
- Keying material for other things, e.g. for QoS admission 
  

It was then decided to divide the security discussion into two separate
areas: 
- Encryption 
- Authentication 
  

2. ENCRYPTION 
============= 
End-to-end encryption only works in the simplest cases. SIP never really
looked at
security from a system perspective, only from a relatively simple
protocol perspective. A
discussion about threats and trust models was therefore initiated. 

There are two types of attacks to worry about: 
- User attacks 
- Rogue proxies 

Rogue proxies are: 
- Proxies that do not do what they are supposed to do, e.g. maintain
privacy 
- Proxies that do things they are not supposed to, e.g. claim that calls
occur when they
didn’t by originating INVITEs. 

The rogue proxy issue led to a basic question; to what extent do you
actually trust that
a proxy does what it’s supposed to ? It was agreed that it would be very
complicated if
you trust the proxy to do some things, but not others, and hence the
model would be to
either trust the proxy or not. 

Furthermore, it was suggested, that trust would be either hop-by-hop or
end-to-end (not,
e.g., trust the next two hops). This led to a decision, that we need a
model that does
not depend on end-to-end security, but allows for it, while still
providing a useful
security service. The model for this will be to have hop-by-hop security
with
transitivity of trust and still allow for end-to-end security. 

As an added requirement on the hop-by-hop security trust model, it was
also decided, that
should a proxy route over an untrusted hop, the caller can request the
call be rejected
rather than continue with the call. 

This decision led to a long discussion about whether it works similarly
the other way,
i.e., can the called party decide to reject a call that has traversed an
untrusted hop.
It was eventually agreed, that it does work similarly the other way.
However, the only
thing you can do when a message has traversed an untrusted hop is to say
that the entire
message is untrusted (e.g., cannot say that certain fields are trusted
and others are
not). 
  

It was agreed to use the above as the foundation for the security model
for encryption. A
discussion on encryption mechanisms then started. 
  

Jonathan Rosenberg opened the discussion by suggesting that: 
- SIP encryption (not authentication) should be deprecated, because: 
  o It doesn’t work with NAT and firewalls 
  o It assumes you know the public key of the party you are calling,
which is not a good
assumption. 
  o It only really hides the body (and again that doesn’t work well due
to the above). 

Initial discussion of these points led to the decision that it should
still be possible
to support end-to-end encryption as there may be some uses for it. 

A problem with the current end-to-end encryption scheme is, that it
relies on knowing the
public key of the other party. One way of solving that is to return a
public key in the
response and simply do a re-INVITE then. However, it was agreed that we
should not do
that, as we effectively end up duplicating IKE, or something similar. 

Dave Oran suggested that end-to-end information that needs encryption
should simply go in
the body (don’t encrypt headers). We could use multipart for this, or
simply allow for a
single body to be encrypted and define an error code to be returned if
the proxy needs to
see the body (e.g. if SDP is encrypted and message traverses a
firewall). The group
decided to accept this mechanism. The error code will only support
coarse granularity,
i.e., if multiple bodies are encrypted, you will not be told which one
is needed in
clear, just a general error saying “body not understood”. 

Additional discussion on encryption of headers ensued. Contact is an
example of header
that can be usefully encrypted (end-to-end), but in general, there
doesn’t seem to be a
whole lot of use for this (generally, if the information is relevant it
can’t be
encrypted). Only want the registrar to see the Contact, not any of the
intermediate
proxies. 

It was then proposed, that rather than having the present encryption
mechanism, we only
encrypt bodies. For those rare SIP headers that you would want to
encrypt end-to-end you
may want to have some kind of “SIP-header” body part encrypted with e.g.
S/MIME (although
S/MIME is not viewed upon favorably by security people). PGP is there
now, but it is not
usefully deployable. 
  

This concluded the encryption discussion with the following recap of the
encryption
requirements identified: 
- Support encryption of some headers (Contact is the one useful example
so far). 
- Prefer encrypted headers (not headers with encrypted content) to end
up in body-parts
rather than in SIP header. 
- Some bodies may have to be encrypted (not necessarily all) 
- Be able to send INVITE without end-to-end security, get some kind of
response back from
remote party telling you to do it securely, and re-issue INVITE (w.
security this time).
Not clear what the exact mechanism is for this, e.g., return public-key,
certificate, or
something outside of SIP. 
- Be able to send INVITE with end-to-end security first time around. 
  

3. AUTHENTICATION (end-to-end) 
============================== 
We have the following potential authentication requirements: 
- Called party id 
- Calling party id 
- Who we thought we were calling 

It is not clear there is any way to achieve this reliably (securely)
unless we have
end-to-end authentication. The problem with end-to-end authentication
is, that it implies
we (a proxy) cannot rewrite any of the information. 

Brian Rosen wanted to have SIP carry some information that
allows/supports end-to-end
authentication. 

Dean Willis wanted to transport messages across untrusted proxies and be
able to detect
what (and how) a message was altered/rewritten. Dean noted that there
are three
possibilities for what can happen to a message: 
- It is not modified 
- It is modified 
- Parts of it are deleted 

It was then suggested that From, To, and Bodies should be authenticated,
however
agreement was not achieved on this. 

Dave Oran then noted that X.400 had solved similar issues and pointed
out the parallels
to headers, bodies and envelopes. It was suggested that we adopt a
similar model. 

This lead to the following tentative decision (not clear if this was in
fact agreed to,
however nobody objected): 
- SIP headers are hop-by-hop (period) 
- Anything above the auth header may be modified hop-by-hop 
- Anything below the auth header is end-to-end (allow for duplication of
fields, e.g. To
and From ) and cannot be modified. 
  

3.1 Hop-by-hop Authentication 
============================= 
Based on the above, the discussion then moved on to focus on hop-by-hop
authentication. 

It was stated that, that we cannot assume availability of public key
cryptography on an
end-system (UA), however we can assume that proxies have public key
cryptography support. 

It was furthermore stated that, we want to differ between security for: 
- Server to server 
- Client to server 

Currently SIP does not specify how you do hop-by-hop security, only that
you can do it.
We want to have at least one mechanism specified in the SIP
specification. 

It was decided that we should have the following hop-by-hop security
mechanisms: 
- Server to server: IPSec using ESP transport, 
- Client to server: IPSec using ESP transport (not clear ESP transport
was agreed to) 

It was then suggested (however Jonathan Rosenberg did not want to
declare consensus yet),
that the following keying/authentication mechanisms should be used: 
- Server to server: Kerberos (with PKCROSS between realms) 
- Client to server: Kerberos (either with shared keys or PKINIT) 
  

The question about whether we need (hop-by-hop) authentication for both
devices and
applications was then asked. It was agreed that we only need it for
applications as we
can always have a default user on the device if no “real” user is logged
in.  The
Security Association established identifies the user. For a gateway, the
“user” is the
gateway. For a device with multiple users on it, the device itself will
then have to
authenticate the users, unless you have a separate Security Association
for every single
user (which you probably don’t want to). 
  

Dean Willis and Robert Sparks objected that you need app-level security
when you don’t
have transitivity of trust, and there are applications where you do not
have transitivity
of trust. For example, consider a call where you involve multiple
providers for that call
(multi-proxy authenticate). Basically, it’s not enough to have either
end-to-end or
hop-by-hop security; you also need to have end-to-intermediate security. 

This was a different model than had been worked on for the majority of
the meeting
however no one argued against the validity of it. Since it contradicts
some of the
earlier assumptions and decisions made during the meeting, each of these
will have to be
revisited. 
  
  
  

4. Miscellaneous 
================ 
In addition to the above, the following security related issues were
raised as well: 

4.1 Areas needing further work 
============================== 
- Challenge-response doesn’t work with forking (but addresses replay) as
only one 401
will get back to the caller. 
- Want to have a security scheme that works with forking (but may not
address replay). 
  

4.2 CANCEL security 
=================== 
- If we have hop-by-hop security and we only accept CANCEL from previous
hop, is CANCEL
security hole then closed ? 
  
  
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Jonathan,

I'm proposing some strawman answers to your questions below...

At 03:40 PM 5/10/00 , Jonathan Rosenberg wrote:
>...
> > Can we at least draw some consensus on the 'Message Waiting
> > Indication(MWI)'
> > event in Voice mail context ?
>
>I think there is a clear need for subscription and notification events
>for many things, including message waiting indication. The question is
>how to structure this in the most general way. I really don't want a
>specific and separate spec just for MWI, as it seems to narrow.
>
> > Further, in one of the threads on this list, Henning says that we must
> > adopt
> > drafts/discussion on PINT to decide on MWI.
>
>To me, what this means is that:
>
>1. we use the SUBSCRIBE and NOTIFY methods
>2. the body is used to define the object being subscribed to

What about an explicit UNSUBSCRIBE method (pint), vs. using SUBSCRIBE with 
an "Expires: 0" header(subscribe/notify draft)?

Why is a body strictly needed if you can express the symantics clearly in 
the Request-URI and an Event header?  for example:

         Event: mailbox-summary

>You should absolutely not be using REGISTER for this.

I wholeheartedly agree!

>Some of the questions in my mind are:
>
>1. how do we handle addressing? I.e., what is the entity that is being
>subscribed to, in the request URI, for mwi? Is it even a SIP URL?

I could imagine any of the following:

sip:rohan@cisco.com
sip:rohan@voicemail-server.cisco.com
sip:rohan@cisco.com?Accept-Contact=*,feature=voice-mail
mailto:rohan@cisco.com  (if my voicemail is stored on my email server)

In the first case, the event relates to a user who has a SIP presence.
In the next, the event relates to my account on my voicemail server.
In the third, the event relates to my "voice-mail" SIP feature.
In the fourth, the event relates to my mail account.

I think that generically this could be left to implementation, but that the 
WG should provide guidelines.  Recommended SIP URI naming conventions in 
general are still pretty undefined.

>2. What is the format of the content in the body for mwi? Is there a
>format which is more generic than just mwi?

Generically, I think we want "message summary" information.

Alternatively, we may want to know only that there are new messages or not 
(simple boolean status, like biff). While this is very simple and good 
enough for some UAs, I don't want this to be the only recommended way to do 
MWI.

A message summary would enable much richer uses, like this message on my 
display:

"You have 3 new, and 7 old voice messages"  or   " *3*/7 voice, *25*/270 
email "

Does anyone else have thoughts on this?

>3. What is the format of the content in notifications?

Manoj, Ilya, and Shail proposed a simple plain-text format.  Does anyone 
have specific complaints about it?

I originally proposed an XML-based format, but it required a much heavier 
parser.

Does anyone have an alternate proposal?

>4. security issues - how do you prevent attacks whereby I subscribe to
>some events, and list someone else in the Contact, say, to receive the
>notifications?

The same way we prevent unauthorized third-parties from REGISTERing with 
someone else in Contact to receive their invites.  ;-)

thanks,
-rohan



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From sip-admin@lists.bell-labs.com  Thu May 11 08:48:24 2000
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I think there could be scope for minor confusion over the intended use
of Expires header and Contact expires param in responses to REGISTER
requests.

In the response to a REGISTER section 4.2.6 of the bis draft states that
an Expires header or expires Contact Param should be used to indicate
the time at which the registration will be dropped. To me this can be
taken to mean the two can be interchangeable and so presumably we are
are talking about the time a new Contact contained in the REGISTER will
be dropped. However, in section 6.22 it says that the Expires header is
used in the the response to a REGISTER to indicate the earliest expiry
time of all registrations.

Furthermore is it correct to say that in the response to a REGISTER,
where active registrations already existed, the Expires header cannot be
used to indicate the time the new registration will be dropped and so an
expire Contact param has to be used. If so what is the point in a server
which adds expires Contact params to all active Contacts also adding an
Expires header to indicate the earliest expiry time of all
registrations.

Cheers,
Neil.
--
Ubiquity Software Corporation           http://www.ubiquity.net



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Jonathan,

I think I was trying to make the same point; perhaps I wasn't direct enough.
I do think that most people, perhaps erroneously, equate URI to
URI-reference with respect to BNF. And the one place that I know about that
has something close to a complete SIP BNF:

<http://www.cs.columbia.edu/~hgs/sip/SIPgrammar.html>

actually links URI (and Request-URI for that matter) to a rule called
URI-reference but is defined as:

URI-reference = ([ absoluteURI | relativeURI] [ "#" fragment]) | (sip-url)

So I think some clarification in this area would be helpful.

Rick


Jonathan Rosenberg wrote:

> Rick Workman wrote:
> >
> >
> >
> > Sean Olson wrote:
> >
> > > > The current SIP BNF allows the general form of URI (or
> > > URI-reference?) to
> > > > be used in the request line and various headers. Observations and
> > > > questions:
> > > >
> > > > 1. A valid form of URI (RFC 2396)is the empty sequence since:
> > > >
> > > > URI-reference = [ absoluteURI | relativeURI ] [ "#" fragment ]
> > > >
> > > > This would permit the request line and various headers (To, From,
> > > > Contact. ..) without an address specification which doesn't seem
> > > to make
> > > > much sense. RFC 2396 specifies that the empty sequence URI refers
> > > to the
> > > > start of the current document; is there any corresponding
> > > abstraction for
> > > > SIP transcations?
> > >
> > > No. Please god no. Obviously RFC2396 specifies a much wider and more
> > > abstract form
> > > of URI than is commonly used in SIP. Not everything in 2396 is
> > > appropriate for
> > > every URI scheme. They are a set of guidelines that work more or
> > > less for the most
> > > common URI schemes in use today: http, ftp, telnet, mailto (with
> > > some quirks), etc.
> > >
> >
> > Be that as it may, they're guidelines which are now an integral part
> > of the SIP BNF.
>
> Actually, this doesn't seem to be the case. The BNF for request URI, for
> example, is never defined in RFC2543! It says in section 4.3 thats its
> either a SIP URI or general URI. The BNF for addr-spec (used in To,
> From, Contact) is defined as SIP-URL or URI, but I also never found the
> BNF for URI. Even rfc2396 doesn't define "URI", but rather absoluteURI,
> relativeURI or URI-reference.
>
> I believe what URI really means in the SIP context is absoluteURI. That
> is, there must always be a scheme, followed by a colon. I believe this
> eliminates many of the ambiguities people have been raising. I'm not a
> big fan of relative URIs within HTTP itself (as opposed to within HTML,
> where the browser computes the absolute URI and puts that in the HTTP
> request); in general, I believe URIs sent on the wire should be globally
> resolvable. HTTP ran into all sorts of problems, as I recall, since URIs
> were not always absolute, and thus servers that supported many domains
> got confused when they received things like GET index.html HTTP/1.0.
>
> >
> > >
> > >
> > > >
> > > > 2. The relative URI references a resource relative to a base URI
> > > of a
> > > > hierarchial scheme. So a similar question: is this a useful
> > > concept for
> > > > SIP sessions?
> > > >
> > >
> > > The only contexts where I would think this might be useful would be
> > > in
> > > 1) The headers parameter of a SIP URI (for the From: and possibly
> > > To: headers)
> > > 2) A SIP "cookie" similar to an HTTP-cookie. This hasn't really been
> > > investigated
> > >    much but would be interesting.
> > >
> > > In other words, I don't think it is generally useful in the usual
> > > places in a SIP
> > > message (To/From/Contact/Request-URI). But it is potentially useful
> > > at the user interface
> > > level or as part of some service logic.
> >
> > As part of a SIP message? I guess I can think of many examples where
> > you want to map user friendly "addresses" to URI's in SIP messages,
> > but that's out of scope for this discussion.
>
> I really don't see the use of relative URIs within SIP itself.
>
> >
> > >
> > >
> > > > 3. Finally RFC 2396 states (my emphasis):
> > > >   "When a URI reference is used to perform a retrieval action on
> > > the
> > > >    identified resource, the optional fragment identifier,
> > > separated from
> > > >    the URI by a crosshatch ("#") character, consists of additional
> > >
> > > >    reference information to be interpreted by the user agent after
> > > the
> > > >    retrieval action has been successfully completed.  As such, it
> > > is not
> > > >    part of a URI, but is often used in conjunction with a URI."
> > > >
> > >
> > > True, but this treated in very different ways by different web
> > > browsers. Some
> > > treat it as part of the URI, others use it for "post-processing".
> > > It's not really used in a standard way for SIP, but as stated above
> > > it really is up
> > > to the UserAgent's interpretation. THIS IS A GOOD THING.
> >
> > I'm not arguing the merits of the URI syntax, just how it's used in
> > SIP messages.
>
> Again, if we say that URI = absoluteURI, this problem goes away also.
>
> -Jonathan R.
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com

--------------AA68DC2E7F58AE2CE0596D11
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<tt>Jonathan,</tt><tt></tt>
<p><tt>I think I was trying to make the same point; perhaps I wasn't direct
enough. I do think that most people, perhaps erroneously, equate URI to
URI-reference with respect to BNF. And the one place that I know about
that has something close to a complete SIP BNF:</tt><tt></tt>
<p><tt>&lt;<A HREF="http://www.cs.columbia.edu/~hgs/sip/SIPgrammar.html">http://www.cs.columbia.edu/~hgs/sip/SIPgrammar.html</A>></tt><tt></tt>
<p><tt>actually links URI (and Request-URI for that matter) to a rule called
URI-reference but is defined as:</tt><tt></tt>
<p><tt>URI-reference = ([ absoluteURI | relativeURI] [ "#" fragment]) |
(sip-url)</tt><tt></tt>
<p><tt>So I think some clarification in this area would be helpful.</tt><tt></tt>
<p><tt>Rick</tt>
<br><tt></tt>&nbsp;<tt></tt>
<p><tt>Jonathan Rosenberg wrote:</tt>
<blockquote TYPE=CITE><tt>Rick Workman wrote:</tt>
<br><tt>></tt>
<br><tt>></tt>
<br><tt>></tt>
<br><tt>> Sean Olson wrote:</tt>
<br><tt>></tt>
<br><tt>> > > The current SIP BNF allows the general form of URI (or</tt>
<br><tt>> > URI-reference?) to</tt>
<br><tt>> > > be used in the request line and various headers. Observations
and</tt>
<br><tt>> > > questions:</tt>
<br><tt>> > ></tt>
<br><tt>> > > 1. A valid form of URI (RFC 2396)is the empty sequence since:</tt>
<br><tt>> > ></tt>
<br><tt>> > > URI-reference = [ absoluteURI | relativeURI ] [ "#" fragment
]</tt>
<br><tt>> > ></tt>
<br><tt>> > > This would permit the request line and various headers (To,
From,</tt>
<br><tt>> > > Contact. ..) without an address specification which doesn't
seem</tt>
<br><tt>> > to make</tt>
<br><tt>> > > much sense. RFC 2396 specifies that the empty sequence URI
refers</tt>
<br><tt>> > to the</tt>
<br><tt>> > > start of the current document; is there any corresponding</tt>
<br><tt>> > abstraction for</tt>
<br><tt>> > > SIP transcations?</tt>
<br><tt>> ></tt>
<br><tt>> > No. Please god no. Obviously RFC2396 specifies a much wider
and more</tt>
<br><tt>> > abstract form</tt>
<br><tt>> > of URI than is commonly used in SIP. Not everything in 2396
is</tt>
<br><tt>> > appropriate for</tt>
<br><tt>> > every URI scheme. They are a set of guidelines that work more
or</tt>
<br><tt>> > less for the most</tt>
<br><tt>> > common URI schemes in use today: http, ftp, telnet, mailto
(with</tt>
<br><tt>> > some quirks), etc.</tt>
<br><tt>> ></tt>
<br><tt>></tt>
<br><tt>> Be that as it may, they're guidelines which are now an integral
part</tt>
<br><tt>> of the SIP BNF.</tt><tt></tt>
<p><tt>Actually, this doesn't seem to be the case. The BNF for request
URI, for</tt>
<br><tt>example, is never defined in RFC2543! It says in section 4.3 thats
its</tt>
<br><tt>either a SIP URI or general URI. The BNF for addr-spec (used in
To,</tt>
<br><tt>From, Contact) is defined as SIP-URL or URI, but I also never found
the</tt>
<br><tt>BNF for URI. Even rfc2396 doesn't define "URI", but rather absoluteURI,</tt>
<br><tt>relativeURI or URI-reference.</tt><tt></tt>
<p><tt>I believe what URI really means in the SIP context is absoluteURI.
That</tt>
<br><tt>is, there must always be a scheme, followed by a colon. I believe
this</tt>
<br><tt>eliminates many of the ambiguities people have been raising. I'm
not a</tt>
<br><tt>big fan of relative URIs within HTTP itself (as opposed to within
HTML,</tt>
<br><tt>where the browser computes the absolute URI and puts that in the
HTTP</tt>
<br><tt>request); in general, I believe URIs sent on the wire should be
globally</tt>
<br><tt>resolvable. HTTP ran into all sorts of problems, as I recall, since
URIs</tt>
<br><tt>were not always absolute, and thus servers that supported many
domains</tt>
<br><tt>got confused when they received things like GET index.html HTTP/1.0.</tt><tt></tt>
<p><tt>></tt>
<br><tt>> ></tt>
<br><tt>> ></tt>
<br><tt>> > ></tt>
<br><tt>> > > 2. The relative URI references a resource relative to a base
URI</tt>
<br><tt>> > of a</tt>
<br><tt>> > > hierarchial scheme. So a similar question: is this a useful</tt>
<br><tt>> > concept for</tt>
<br><tt>> > > SIP sessions?</tt>
<br><tt>> > ></tt>
<br><tt>> ></tt>
<br><tt>> > The only contexts where I would think this might be useful
would be</tt>
<br><tt>> > in</tt>
<br><tt>> > 1) The headers parameter of a SIP URI (for the From: and possibly</tt>
<br><tt>> > To: headers)</tt>
<br><tt>> > 2) A SIP "cookie" similar to an HTTP-cookie. This hasn't really
been</tt>
<br><tt>> > investigated</tt>
<br><tt>> >&nbsp;&nbsp;&nbsp; much but would be interesting.</tt>
<br><tt>> ></tt>
<br><tt>> > In other words, I don't think it is generally useful in the
usual</tt>
<br><tt>> > places in a SIP</tt>
<br><tt>> > message (To/From/Contact/Request-URI). But it is potentially
useful</tt>
<br><tt>> > at the user interface</tt>
<br><tt>> > level or as part of some service logic.</tt>
<br><tt>></tt>
<br><tt>> As part of a SIP message? I guess I can think of many examples
where</tt>
<br><tt>> you want to map user friendly "addresses" to URI's in SIP messages,</tt>
<br><tt>> but that's out of scope for this discussion.</tt><tt></tt>
<p><tt>I really don't see the use of relative URIs within SIP itself.</tt><tt></tt>
<p><tt>></tt>
<br><tt>> ></tt>
<br><tt>> ></tt>
<br><tt>> > > 3. Finally RFC 2396 states (my emphasis):</tt>
<br><tt>> > >&nbsp;&nbsp; "When a URI reference is used to perform a retrieval
action on</tt>
<br><tt>> > the</tt>
<br><tt>> > >&nbsp;&nbsp;&nbsp; identified resource, the optional fragment
identifier,</tt>
<br><tt>> > separated from</tt>
<br><tt>> > >&nbsp;&nbsp;&nbsp; the URI by a crosshatch ("#") character,
consists of additional</tt>
<br><tt>> ></tt>
<br><tt>> > >&nbsp;&nbsp;&nbsp; reference information to be interpreted
by the user agent after</tt>
<br><tt>> > the</tt>
<br><tt>> > >&nbsp;&nbsp;&nbsp; retrieval action has been successfully
completed.&nbsp; As such, it</tt>
<br><tt>> > is not</tt>
<br><tt>> > >&nbsp;&nbsp;&nbsp; part of a URI, but is often used in conjunction
with a URI."</tt>
<br><tt>> > ></tt>
<br><tt>> ></tt>
<br><tt>> > True, but this treated in very different ways by different
web</tt>
<br><tt>> > browsers. Some</tt>
<br><tt>> > treat it as part of the URI, others use it for "post-processing".</tt>
<br><tt>> > It's not really used in a standard way for SIP, but as stated
above</tt>
<br><tt>> > it really is up</tt>
<br><tt>> > to the UserAgent's interpretation. THIS IS A GOOD THING.</tt>
<br><tt>></tt>
<br><tt>> I'm not arguing the merits of the URI syntax, just how it's used
in</tt>
<br><tt>> SIP messages.</tt><tt></tt>
<p><tt>Again, if we say that URI = absoluteURI, this problem goes away
also.</tt><tt></tt>
<p><tt>-Jonathan R.</tt>
<br><tt>--</tt>
<br><tt>Jonathan D. Rosenberg&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
72 Eagle Rock Ave.</tt>
<br><tt>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</tt>
<br><tt>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</tt>
<br><tt>jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
FAX:&nbsp;&nbsp; (732) 741-4778</tt>
<br><tt><a href="http://www.cs.columbia.edu/~jdrosen">http://www.cs.columbia.edu/~jdrosen</a>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
PHONE: (732) 741-7244</tt>
<br><tt><a href="http://www.dynamicsoft.com">http://www.dynamicsoft.com</a></tt></blockquote>
<tt></tt></html>

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From sip-admin@lists.bell-labs.com  Thu May 11 10:51:13 2000
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From: Gethin Liddell <gethin@ubiquity.net>
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Subject: [SIP] Another grammar query - Call-ID
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Hi all,

I have another grammar query, this time regarding the call-id.  

Call-ID is defined as (bis 6.12):

callid = local-id  @  host
local-id = 1*uric
Call-ID = (  Call-ID  |  i  )  :  callid

and uric is defined as (bis Section 2 Figure 3):

uric = reserved | unreserved | escaped
reserved =  ;  |  /  |  ?  |  :  |  @  |  & |  =  |  +  |  $  |  ,  |  [  |  ] 

The problem here is with @.

a valid local-id is 

a@b

but that is also a valid 

local-id @ host

so which is it?
A parser is likely to claim that this is a local-id with no "@ host" value and declare 
it an invalid call-id.

I think this is the same problem that occured with the definition of the Sip-URL and 
escaped characters in Section 2. Am i right in thinking that what was actually meant 
was:

uric = unreserved | escaped

where escaped allows the characters in the reserved set to be included?  

Thus any processing (equality checking etc...) of local-id is done on the un-escaped 
version but when transmitted in the message the escaped form must be used.  If this is 
the case, I don't see any reason why any of the other characters in the reserved set 
need to be escaped.

Quoting section 2:

"Note that reserved characters have to be escaped and that the `set of characters 
reserved within any given URI component is defined by that component. In general, a 
character is reserved if the semantics of the URI changes if the character is replaced 
with its escaped US-ASCII encoding' [12]."

Applying this to the call-id, only the @ changes the semantics of the Call-ID and so
only the @ needs to be escaped. (Of course it is the case that any character in the 
local-id MAY be escaped and this MUST be handled by the parser).

Am i correct about this?  If so I think the whole `escaped' issue needs to be clarified
somewhere (or have i just missed where it is clairified).

regards 

gethin


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Subject: Re: [SIP] Another grammar query - Call-ID
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> 
> Hi all,
> 
> I have another grammar query, this time regarding the call-id.  
> 
> Call-ID is defined as (bis 6.12):
> 
> callid = local-id  @  host
> local-id = 1*uric
> Call-ID = (  Call-ID  |  i  )  :  callid
> 

I may be wrong, but I seem to remember some agreement that Call-ID would become
a token and would not necessarily need to take the form "local-id@host".
But perhaps more importantly, I think its dangerous to start trying to parse the Call-ID
and possibly derive information from it, such as the host. 

--
Sean Olson <sean.olson@ericsson.com>


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From: Gethin Liddell <gethin@ubiquity.net>
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To: Sean Olson <eussean@exu.ericsson.se>, sip@lists.bell-labs.com
Subject: Re: [SIP] Another grammar query - Call-ID
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On Thu, 11 May 2000, Sean Olson wrote:
> > 
> > Hi all,
> > 
> > I have another grammar query, this time regarding the call-id.  
> > 
> > Call-ID is defined as (bis 6.12):
> > 
> > callid = local-id  @  host
> > local-id = 1*uric
> > Call-ID = (  Call-ID  |  i  )  :  callid
> > 
> 
> I may be wrong, but I seem to remember some agreement that Call-ID would become
> a token and would not necessarily need to take the form "local-id@host".

Great, that makes things easier :)

Can this be confirmed?

> But perhaps more importantly, I think its dangerous to start trying to parse the Call-ID
> and possibly derive information from it, such as the host. 


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To: Gethin Liddell <gethin@ubiquity.net>
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Subject: Re: [SIP] Another grammar query - Call-ID
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Gethin Liddell wrote:
> 
> On Thu, 11 May 2000, Sean Olson wrote:
> > >
> > > Hi all,
> > >
> > > I have another grammar query, this time regarding the call-id.
> > >
> > > Call-ID is defined as (bis 6.12):
> > >
> > > callid = local-id  @  host
> > > local-id = 1*uric
> > > Call-ID = (  Call-ID  |  i  )  :  callid
> > >
> >
> > I may be wrong, but I seem to remember some agreement that Call-ID would become
> > a token and would not necessarily need to take the form "local-id@host".

Yes, except token didn't work because it doesn't include the '@'
character. Basically, the field should be treated as an opaque,
uninterpreted identifier, e.g. TEXT-UTF8-TRIM in the grammar.

> 
> Great, that makes things easier :)
> 
> Can this be confirmed?
> 
> > But perhaps more importantly, I think its dangerous to start trying to parse the Call-ID
> > and possibly derive information from it, such as the host.
> 

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


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From sip-admin@lists.bell-labs.com  Thu May 11 11:45:33 2000
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Date: Thu, 11 May 2000 16:46:23 +0100
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To: Neil Deason <ndeason@ubiquity.net>
Cc: sip@lists.bell-labs.com
Subject: Re: [SIP] Expires header vs expires Contact param
References: <391AAB8B.B29E7C36@ubiquity.net>
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My understanding of this is that the Expires header, if present, applies
to all Contacts in a REGISTER request or response, UNLESS an expires
parameter is present for that Contact, in which case the expires param
takes precedence as it is more specific.

Anders


Neil Deason wrote:
> 
> I think there could be scope for minor confusion over the intended use
> of Expires header and Contact expires param in responses to REGISTER
> requests.
> 
> In the response to a REGISTER section 4.2.6 of the bis draft states that
> an Expires header or expires Contact Param should be used to indicate
> the time at which the registration will be dropped. To me this can be
> taken to mean the two can be interchangeable and so presumably we are
> are talking about the time a new Contact contained in the REGISTER will
> be dropped. However, in section 6.22 it says that the Expires header is
> used in the the response to a REGISTER to indicate the earliest expiry
> time of all registrations.
> 
> Furthermore is it correct to say that in the response to a REGISTER,
> where active registrations already existed, the Expires header cannot be
> used to indicate the time the new registration will be dropped and so an
> expire Contact param has to be used. If so what is the point in a server
> which adds expires Contact params to all active Contacts also adding an
> Expires header to indicate the earliest expiry time of all
> registrations.
> 
> Cheers,
> Neil.
> --
> Ubiquity Software Corporation           http://www.ubiquity.net
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

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


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Date: Thu, 11 May 2000 11:30:54 -0500
From: Dean Willis <dean.willis@wcom.com>
In-reply-to: <3919E578.9FCE6061@dynamicsoft.com>
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Subject: [SIP] Subscription and Notification Concepts (was RE: Two New Drafts)
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Just as a reference point, there is an excellent paper on the Cambridge
Event Architecture in the March 2000 "Computer". This included discussion of
general issues around asynchronous event classification, roles, and use of
certificates for identification and authorization in a role-membership
model.

While the implementation may be heavily IPRed (I believe Nortel underwote
some of the work), I think some of the general concepts and nomenclature
would be helpful to our discussion.

I don't have the time at the moment to digest it fully, but it may be that
some of our more studiouscompanions might wish to review this material and
report on its applicability.

--
Dean


> -----Original Message-----
> From: sip-admin@lists.bell-labs.com
> [mailto:sip-admin@lists.bell-labs.com]On Behalf Of Jonathan Rosenberg
> Sent: Wednesday, May 10, 2000 5:41 PM
> To: Manoj Bhatia
> Cc: archow@hss.hns.com; Adam B. Roach; sip@lists.research.bell-labs.com;
> stlevy@cisco.com
> Subject: [SIP] Re: Two New Drafts
>
>
>
>
> Manoj Bhatia wrote:
> >
> > Hi Jonathan
> >
> > Can we at least draw some consensus on the 'Meesage Waiting
> > Indication(MWI)'
> > event in Voice mail context ?
>
> I think there is a clear need for subscription and notification events
> for many things, including message waiting indication. The question is
> how to structure this in the most general way. I really don't want a
> specific and separate spec just for MWI, as it seems to narrow.
>
> >
> > Further, in one of the threads on this list, Henning says that we must
> > adopt
> > drafts/discussion on PINT to decide on MWI.
>
> To me, what this means is that:
>
> 1. we use the SUBSCRIBE and NOTIFY methods
> 2. the body is used to define the object being subscribed to
>
>
> >
> > Also, there is some concern that whether SUBSCRIBE -  a new SIP method
> >
> > should be used for sending subscription requests or is REGISTER a good
> > way of doing that ?
>
> Definitely not REGISTER. Its semantics are completely different. It
> updates a binding, in a server, of the address in the To with the
> addresses in the Contact. I don't see at all how this accomplishes MWI.
>
> >
> > My concern is that very soon you may have interop problems with
> > SUBSCRIBE/
> > REGISTER method being used by vendors in different ways .
>
> You should absolutely not be using REGISTER for this.
>
> >
> > Is SUBSCRIBE/NOTIFY the de facto way for MWI implementation ?
>
> I think it is clearly the basis. The question, again, is how to do this
> generally.
>
> Some of the questions in my mind are:
>
> 1. how do we handle addressing? I.e., what is the entity that is being
> subscribed to, in the request URI, for mwi? Is it even a SIP URL?
>
> 2. What is the format of the content in the body for mwi? Is there a
> format which is more generic than just mwi?
>
> 3. What is the format of the content in notifications?
>
> 4. security issues - how do you prevent attacks whereby I subscribe to
> some events, and list someone else in the Contact, say, to receive the
> notifications?
>



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I started an informal mailing list to hash out some of the details of the 
SUBSCRIBE/NOTIFY extension to SIP. You can find this at:

http://www.egroups.com/group/sip-sub

I don't intend to subvert the main SIP mailing list, this is just a place to hash
out some ideas until the SIP working group decides to make this a work item. 

For a starting point, take a look at the following drafts:

http://www.ietf.org/internet-drafts/draft-roach-sip-subscribe-notify-00.txt

http://www.cs.columbia.edu/sip/drafts/draft-rosenberg-sip-pip-00.txt

and

http://www.ietf.org/internet-drafts/draft-ietf-pint-protocol-04.txt

--
Sean Olson <sean.olson@ericsson.com>
Ericsson Inc.
tel: +1-972-583-5472


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From sip-admin@lists.bell-labs.com  Thu May 11 13:26:52 2000
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Subject: Re: [SIP] Expires header vs expires Contact param
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Anders Kristensen wrote:

> My understanding of this is that the Expires header, if present, applies
> to all Contacts in a REGISTER request or response, UNLESS an expires
> parameter is present for that Contact, in which case the expires param
> takes precedence as it is more specific.

Thanks. Looking again 6.13 of the bis draft does state that in a REGISTER 2xx
response if a Contact entry does not have an optional expires param, the value
of the Expires header is used. If neither mechanism is used, the expiration
time specified in the request, explicitly or default, is used.

Is this compatible with using the Expires header to indicate the earliest
expiry time of all registrations (6.22). Imagine a user registers from a
location supplying no expiry time. The server assumes 1 hour and returns a 2xx
response which includes this new registration and another longstanding
registration. Isn't it possible that the new Contact could contain no expires
param and the Expires header be set to the duration of the long standing
registration?

Is there a reason why servers shouldn't be made to use the expires Contact
param for every Contact? This would avoid the potential for any confusion and
make the need for the Expires header to state the earliest expiry time of all
registratations largely redundant.

Cheers,
Neil
--
Ubiquity Software Corporation           http://www.ubiquity.net

> > I think there could be scope for minor confusion over the intended use
> > of Expires header and Contact expires param in responses to REGISTER
> > requests.
> >
> > In the response to a REGISTER section 4.2.6 of the bis draft states that
> > an Expires header or expires Contact Param should be used to indicate
> > the time at which the registration will be dropped. To me this can be
> > taken to mean the two can be interchangeable and so presumably we are
> > are talking about the time a new Contact contained in the REGISTER will
> > be dropped. However, in section 6.22 it says that the Expires header is
> > used in the the response to a REGISTER to indicate the earliest expiry
> > time of all registrations.
> >
> > Furthermore is it correct to say that in the response to a REGISTER,
> > where active registrations already existed, the Expires header cannot be
> > used to indicate the time the new registration will be dropped and so an
> > expire Contact param has to be used. If so what is the point in a server
> > which adds expires Contact params to all active Contacts also adding an
> > Expires header to indicate the earliest expiry time of all
> > registrations.



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From: Shail Bhatnagar <shbhatna@cisco.com>
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To: Neil Deason <ndeason@ubiquity.net>
Cc: Anders Kristensen <ak@hplb.hpl.hp.com>, sip@lists.bell-labs.com
Subject: Re: [SIP] Expires header vs expires Contact param
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I think the registrar should return the expires parameter in each
Contact in
the 200 OK, specifying the GMT time at which this
registration/Contact will expire. The Expires header in the 200 OK would
specify the earliest of these expiration time(s).

Comments ??
Shail

Neil Deason wrote:

> Thanks. Looking again 6.13 of the bis draft does state that in a REGISTER 2xx
> response if a Contact entry does not have an optional expires param, the value
> of the Expires header is used. If neither mechanism is used, the expiration
> time specified in the request, explicitly or default, is used.
> 
> Is this compatible with using the Expires header to indicate the earliest
> expiry time of all registrations (6.22). Imagine a user registers from a
> location supplying no expiry time. The server assumes 1 hour and returns a 2xx
> response which includes this new registration and another longstanding
> registration. Isn't it possible that the new Contact could contain no expires
> param and the Expires header be set to the duration of the long standing
> registration?
> 
> Is there a reason why servers shouldn't be made to use the expires Contact
> param for every Contact? This would avoid the potential for any confusion and
> make the need for the Expires header to state the earliest expiry time of all
> registratations largely redundant.
> 
> Cheers,
> Neil
> --
> Ubiquity Software Corporation           http://www.ubiquity.net
> 
> > > I think there could be scope for minor confusion over the intended use
> > > of Expires header and Contact expires param in responses to REGISTER
> > > requests.
> > >
> > > In the response to a REGISTER section 4.2.6 of the bis draft states that
> > > an Expires header or expires Contact Param should be used to indicate
> > > the time at which the registration will be dropped. To me this can be
> > > taken to mean the two can be interchangeable and so presumably we are
> > > are talking about the time a new Contact contained in the REGISTER will
> > > be dropped. However, in section 6.22 it says that the Expires header is
> > > used in the the response to a REGISTER to indicate the earliest expiry
> > > time of all registrations.
> > >
> > > Furthermore is it correct to say that in the response to a REGISTER,
> > > where active registrations already existed, the Expires header cannot be
> > > used to indicate the time the new registration will be dropped and so an
> > > expire Contact param has to be used. If so what is the point in a server
> > > which adds expires Contact params to all active Contacts also adding an
> > > Expires header to indicate the earliest expiry time of all
> > > registrations.
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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Date: Thu, 11 May 2000 13:47:11 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Sean Olson <eussean@exu.ericsson.se>
Cc: sip@lists.research.bell-labs.com, pint@lists.bell-labs.com
Subject: Re: [SIP] Mailing list for SUBSCRIBE/NOTIFY
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Well, I had already created (and I think announced) an egroups list for
it - sip-events@egroups.com. We definitely don't need 2.

-Jonathan R.

Sean Olson wrote:
> 
> I started an informal mailing list to hash out some of the details of the
> SUBSCRIBE/NOTIFY extension to SIP. You can find this at:
> 
> http://www.egroups.com/group/sip-sub
> 
> I don't intend to subvert the main SIP mailing list, this is just a place to hash
> out some ideas until the SIP working group decides to make this a work item.
> 
> For a starting point, take a look at the following drafts:
> 
> http://www.ietf.org/internet-drafts/draft-roach-sip-subscribe-notify-00.txt
> 
> http://www.cs.columbia.edu/sip/drafts/draft-rosenberg-sip-pip-00.txt
> 
> and
> 
> http://www.ietf.org/internet-drafts/draft-ietf-pint-protocol-04.txt
> 
> --
> Sean Olson <sean.olson@ericsson.com>
> Ericsson Inc.
> tel: +1-972-583-5472
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Subject: Re: [SIP] Expires header vs expires Contact param
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Neil Deason wrote:
> 
> Anders Kristensen wrote:
> 
> > My understanding of this is that the Expires header, if present, applies
> > to all Contacts in a REGISTER request or response, UNLESS an expires
> > parameter is present for that Contact, in which case the expires param
> > takes precedence as it is more specific.
> 
> Thanks. Looking again 6.13 of the bis draft does state that in a REGISTER 2xx
> response if a Contact entry does not have an optional expires param, the value
> of the Expires header is used. If neither mechanism is used, the expiration
> time specified in the request, explicitly or default, is used.
> 
> Is this compatible with using the Expires header to indicate the earliest
> expiry time of all registrations (6.22).

Yes. The header has clear meaning. Its only confusing if you interpret
it to mean the expiration time of all Contact headers, rather than the
earliest expiry.

 Imagine a user registers from a
> location supplying no expiry time. The server assumes 1 hour and returns a 2xx
> response which includes this new registration and another longstanding
> registration. Isn't it possible that the new Contact could contain no expires
> param and the Expires header be set to the duration of the long standing
> registration?

Then the registrar screwed up. The expires parameter and Expires header
have clear definitions in the response. In this example, the registrar
set it wrong.

> 
> Is there a reason why servers shouldn't be made to use the expires Contact
> param for every Contact? This would avoid the potential for any confusion and
> make the need for the Expires header to state the earliest expiry time of all
> registratations largely redundant.

I would definitely recommend using the expires parameters for every
contact header. Its an implementation decision, though. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Subject: Re: [SIP] URI's in SIP messages
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Rick Workman wrote:
> 
> Jonathan,
> 
> I think I was trying to make the same point; perhaps I wasn't direct
> enough. I do think that most people, perhaps erroneously, equate URI
> to URI-reference with respect to BNF. And the one place that I know
> about that has something close to a complete SIP BNF:
> 
> <http://www.cs.columbia.edu/~hgs/sip/SIPgrammar.html>
> 
> actually links URI (and Request-URI for that matter) to a rule called
> URI-reference but is defined as:
> 
> URI-reference = ([ absoluteURI | relativeURI] [ "#" fragment]) |
> (sip-url)
> 
> So I think some clarification in this area would be helpful.

Right; I'm basically saying I believe this grammar to be wrong. URI
should be absoluteURI.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Thu May 11 14:02:40 2000
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I don't see any issue with this syntax. Do a reverse search on '@' sign
in
the Call-ID header value - anything preceeding it is the local-id and 
anything after it is the host. I don't think host names can have '@'
sign.


Gethin Liddell wrote:
> 
> Hi all,
> 
> I have another grammar query, this time regarding the call-id.
> 
> Call-ID is defined as (bis 6.12):
> 
> callid = local-id  @  host
> local-id = 1*uric
> Call-ID = (  Call-ID  |  i  )  :  callid
> 
> and uric is defined as (bis Section 2 Figure 3):
> 
> uric = reserved | unreserved | escaped
> reserved =  ;  |  /  |  ?  |  :  |  @  |  & |  =  |  +  |  $  |  ,  |  [  |  ]
> 
> The problem here is with @.
> 
> a valid local-id is
> 
> a@b
> 
> but that is also a valid
> 
> local-id @ host
> 
> so which is it?
> A parser is likely to claim that this is a local-id with no "@ host" value and declare
> it an invalid call-id.
> 
> I think this is the same problem that occured with the definition of the Sip-URL and
> escaped characters in Section 2. Am i right in thinking that what was actually meant
> was:
> 
> uric = unreserved | escaped
> 
> where escaped allows the characters in the reserved set to be included?
> 
> Thus any processing (equality checking etc...) of local-id is done on the un-escaped
> version but when transmitted in the message the escaped form must be used.  If this is
> the case, I don't see any reason why any of the other characters in the reserved set
> need to be escaped.
> 
> Quoting section 2:
> 
> "Note that reserved characters have to be escaped and that the `set of characters
> reserved within any given URI component is defined by that component. In general, a
> character is reserved if the semantics of the URI changes if the character is replaced
> with its escaped US-ASCII encoding' [12]."
> 
> Applying this to the call-id, only the @ changes the semantics of the Call-ID and so
> only the @ needs to be escaped. (Of course it is the case that any character in the
> local-id MAY be escaped and this MUST be handled by the parser).
> 
> Am i correct about this?  If so I think the whole `escaped' issue needs to be clarified
> somewhere (or have i just missed where it is clairified).
> 
> regards
> 
> gethin
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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Subject: Re: [SIP] Expires header vs expires Contact param
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Jonathan Rosenberg wrote:

> Neil Deason wrote:
> >
> > Anders Kristensen wrote:
> >
> > > My understanding of this is that the Expires header, if present, applies
> > > to all Contacts in a REGISTER request or response, UNLESS an expires
> > > parameter is present for that Contact, in which case the expires param
> > > takes precedence as it is more specific.
> >
> > Thanks. Looking again 6.13 of the bis draft does state that in a REGISTER 2xx
> > response if a Contact entry does not have an optional expires param, the value
> > of the Expires header is used. If neither mechanism is used, the expiration
> > time specified in the request, explicitly or default, is used.
> >
> > Is this compatible with using the Expires header to indicate the earliest
> > expiry time of all registrations (6.22).
>
> Yes. The header has clear meaning. Its only confusing if you interpret
> it to mean the expiration time of all Contact headers, rather than the
> earliest expiry.

Thanks Jonathan, this was the source of my confusion. What is the thinking behind
setting the Expires header to the time when the first of a users active registration
runs out. I had mistakenly thought it was the time they all ran out as this would be
significant in that then the user wouldn't be registered anywhere.

Cheers,
Neil.
--
Ubiquity Software Corporation          http://www.ubiquity.net

>
>
>  Imagine a user registers from a
> > location supplying no expiry time. The server assumes 1 hour and returns a 2xx
> > response which includes this new registration and another longstanding
> > registration. Isn't it possible that the new Contact could contain no expires
> > param and the Expires header be set to the duration of the long standing
> > registration?
>
> Then the registrar screwed up. The expires parameter and Expires header
> have clear definitions in the response. In this example, the registrar
> set it wrong.
>
> >
> > Is there a reason why servers shouldn't be made to use the expires Contact
> > param for every Contact? This would avoid the potential for any confusion and
> > make the need for the Expires header to state the earliest expiry time of all
> > registratations largely redundant.
>
> I would definitely recommend using the expires parameters for every
> contact header. Its an implementation decision, though.
>
> -Jonathan R.
>
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com



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My apologies for creating a duplicate list. I somehow missed Jonathon's announcement.
I defer to Jonathan's list :) (http://www.egroups.com/group/sip-events) 

Thanks Jonathan.

--
Sean Olson <sean.olson@ericsson.com>
Ericsson Inc.
tel: +1-972-583-5472 


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Reply-To: James Kempf <James.Kempf@Eng.Sun.COM>
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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Couple of comments.


>3. AUTHENTICATION (end-to-end) 
>============================== 
>We have the following potential authentication requirements: 
>- Called party id 
>- Calling party id 
>- Who we thought we were calling 
>

This doesn't say anything about a client authenticating a proxy,
but that seems to be implied by the subsequent discussion.

>Dean Willis and Robert Sparks objected that you need app-level security
>when you don’t
>have transitivity of trust, and there are applications where you do not
>have transitivity
>of trust. For example, consider a call where you involve multiple
>providers for that call
>(multi-proxy authenticate). Basically, it’s not enough to have either
>end-to-end or
>hop-by-hop security; you also need to have end-to-intermediate security. 
>
>This was a different model than had been worked on for the majority of
>the meeting
>however no one argued against the validity of it. Since it contradicts
>some of the
>earlier assumptions and decisions made during the meeting, each of these
>will have to be
>revisited. 
>  

I`m not exactly sure what is meant by "application level security" here.
Does "application" mean "application built on top of SIP" or does
it mean "SIP as an application protocol?

Certainly in the case of cellular, it will be the case that the
proxy will be changing. When the mobile terminal roams outside
of it's cellular provider's domain, the new proxy needs to be authenticated
with the mobile terminal, and vice versa. This roaming can happen either 
statically or dynamically, i.e. statically when the mobile terminal comes up
in a foreign domain, or dynamically, when a call is in progress
and the mobile roams into an area where its provider does not
have coverage but a provider with a roaming agreement does.

		jak



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From: James Yu <james.yu@neustar.com>
To: jdrosen@dynamicsoft.com
Cc: "'Vance Shipley'" <vances@motivity.ca>,
        John Mainwaring <crm312a@nortelnetworks.com>,
        Mark Foster <mark.foster@neustar.com>,
        iptel@lists.research.bell-labs.com, sip@lists.research.bell-labs.com
Subject: RE: [SIP] Re: TRIP
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It is fine with me.

James

> -----Original Message-----
> From: S_S_Dinesh/India/IBM@au1.ibm.com
> [mailto:S_S_Dinesh/India/IBM@au1.ibm.com]
> Sent: Monday, May 08, 2000 4:40 AM
> To: James Yu
> Cc: 'Vance Shipley'; John Mainwaring; Mark Foster;
> iptel@lists.research.bell-labs.com; sip@lists.research.bell-labs.com
> Subject: [SIP] Re: TRIP
> 
> 
> 
> 
> We've discussed this on the list. Several folks pointed out that
> European systems support overlap, and it was decided that it was not
> reasonable to mandate the originating gateway to do digit 
> collection to
> convert this to en-bloc, although it could be optionally done.
> 
> -Jonathan R.
> 
> James Yu wrote:
> >
> > Vance,
> >
> > Thanks for the explanation.
> >
> > The key questions now are whether overlap signaling (e.g., Feature
> Group-D
> > over Multi-Frequency facility) is still used somewhere and 
> if used where
> the
> > digit collection/enbloc signaling is performed in the existing SCN.
> >
> > If overlap signaling is still used (someone please confirm this and
> indicate
> > where it is still used) and a call is to be routed from the 
> originating
> SCN
> > to the Internet, I guess that the originating GW 
> (SCN-Internet)probably
> has
> > to perform the digit collection/enbloc signaling function 
> so that overlap
> > signaling does not propagate further into the Internet and/or the
> > destination GW/SCN.  Otherwise, some entities in the Internet must
> perform
> > that function.
> >
> > James
> >
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 
> ---------
> This message came from the IETF IPTEL Working Group Mailing List.
> 
> 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 


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Date: Thu, 11 May 2000 19:34:53 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: James Yu <james.yu@neustar.com>
Cc: "'Vance Shipley'" <vances@motivity.ca>,
        John Mainwaring <crm312a@nortelnetworks.com>,
        Mark Foster <mark.foster@neustar.com>,
        iptel@lists.research.bell-labs.com, sip@lists.research.bell-labs.com
Subject: Re: [SIP] Re: TRIP
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FYI, folks, please ignore all email you saw from
S_S_Dinesh/India/IBM@aul.ibm.com. This mail is all resends of old
messages back to the list based on misconfiguration of a machine within
IBM.

-Jonathan R.

James Yu wrote:
> 
> It is fine with me.
> 
> James
> 
> > -----Original Message-----
> > From: S_S_Dinesh/India/IBM@au1.ibm.com
> > [mailto:S_S_Dinesh/India/IBM@au1.ibm.com]
> > Sent: Monday, May 08, 2000 4:40 AM
> > To: James Yu
> > Cc: 'Vance Shipley'; John Mainwaring; Mark Foster;
> > iptel@lists.research.bell-labs.com; sip@lists.research.bell-labs.com
> > Subject: [SIP] Re: TRIP
> >
> >
> >
> >
> > We've discussed this on the list. Several folks pointed out that
> > European systems support overlap, and it was decided that it was not
> > reasonable to mandate the originating gateway to do digit
> > collection to
> > convert this to en-bloc, although it could be optionally done.
> >
> > -Jonathan R.
> >
> > James Yu wrote:
> > >
> > > Vance,
> > >
> > > Thanks for the explanation.
> > >
> > > The key questions now are whether overlap signaling (e.g., Feature
> > Group-D
> > > over Multi-Frequency facility) is still used somewhere and
> > if used where
> > the
> > > digit collection/enbloc signaling is performed in the existing SCN.
> > >
> > > If overlap signaling is still used (someone please confirm this and
> > indicate
> > > where it is still used) and a call is to be routed from the
> > originating
> > SCN
> > > to the Internet, I guess that the originating GW
> > (SCN-Internet)probably
> > has
> > > to perform the digit collection/enbloc signaling function
> > so that overlap
> > > signaling does not propagate further into the Internet and/or the
> > > destination GW/SCN.  Otherwise, some entities in the Internet must
> > perform
> > > that function.
> > >
> > > James
> > >
> > --
> > Jonathan D. Rosenberg                       200 Executive Drive
> > Chief Scientist                             Suite 120
> > dynamicsoft                                 West Orange, NJ 07052
> > jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> > http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> > http://www.dynamicsoft.com
> >
> > ---------
> > This message came from the IETF IPTEL Working Group Mailing List.
> >
> >
> >
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> >

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Didn't get this quite right. RFC 2732 (IPv6 Literal Addresses in URL's)
updates RFC 2396 to include "[" and "]" in the URL reserved set.
Corrected ABNF is:

headers     = "?" header *("&" header)
header      = hname "=" hvalue
hname       = 1*(hname_char)
hvalue      = *(hvalue_char)

hname_char  = alphanum | "-" | "." | "!" | "*" | "_" | "+" | "'" | "~"

hvalue_char = hvreserved | unreserved | escaped
hvreserved  = ";" | "/" | "?" | ":" | "@" | "=" | "+" | "$" | "[" | "]"



Rick Workman
Nortel Networks
Ottawa


>
> In an attempt to get some closure on the SIP-URL header BNF, how
> about:
>
> headers     = "?" header *("&" header)
> header      = hname "=" hvalue
> hname       = 1*(hname_char)
> hvalue      = *(hvalue_char)
>
> hname_char  = alphanum | "-" | "." | "!" | "*" | "_" | "+" | "'" | "~"
>
> hvalue_char = hvreserved | unreserved | escaped
> hvreserved  = ";" | "/" | "?" | ":" | "@" | "=" | "+" |"$"
>
>
> Notes:
> 1. This isn't strictly backward compatible but I can't see any
> practical issues. (Same as proposed change to other-param?)
> 2. hname_char set is a subset of token (to be consistent with
> "field-name") which doesn't include "%" (URI "delims") or "`" (URI
> "unwise").
> 3. hvreserved is a subset of the current reserved rule eliminating
> "&", "," as per earlier discussion, and "[" and "]" (both URI
> "unwise").
> 4. Scanning the current header definitions, at least the following
> must be escaped:
>    - any required LWS, e.g., for token separation or in rfc1123 dates
>    - double quotes in quoted strings
>    - "\" in quoted-pair (comments and quoted-strings)
>    - "%" and "`" in tokens
>    - "<" and ">" in name-addrs
>    - "&" in any embedded SIP-URL headers
>    - "," in # lists
>    - any illegal characters in TEXT-UTF8
>    - any others?
> 5. Multi-byte UTF-8 characters will be represent by a 2*escaped ,
> i.e., the UTF-8 character CA98 will escape to %CA%98. (I think this
> was the intent of Jonathan Lennox's previous proposal.)
>
>
> Also, I'd like to see the text concerning reserved characters and
> escaping clarified. If the BNF can be tightened up to the point where
> the allowable characters for any SIP-URL component have been clearly
> defined, then, by definition, any other characters must be escaped.
> Calling these other characters reserved confuses them with those in
> the "reserved" rule in the URI spec (and the current SIP RFC). Am I
> missing something?
>
>
> Rick Workman
> Nortel Networks
> Ottawa

--------------AC8570DA29B18D948607FC59
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<tt>Didn't get this quite right. RFC 2732 (IPv6 Literal Addresses in URL's)
updates RFC 2396 to include "[" and "]" in the URL reserved set. Corrected
ABNF is:</tt><tt></tt>
<p><tt>headers&nbsp;&nbsp;&nbsp;&nbsp; = "?" header *("&amp;" header)</tt>
<br><tt>header&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = hname "=" hvalue</tt>
<br><tt>hname&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = 1*(hname_char)</tt>
<br><tt>hvalue&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = *(hvalue_char)</tt><tt></tt>
<p><tt>hname_char&nbsp; = alphanum | "-" | "." | "!" | "*" | "_" | "+"
| "'" | "~"</tt><tt></tt>
<p><tt>hvalue_char = hvreserved | unreserved | escaped</tt>
<br><tt>hvreserved&nbsp; = ";" | "/" | "?" | ":" | "@" | "=" | "+" | "$"
| "[" | "]"</tt>
<br><tt>&nbsp;</tt>
<br><tt></tt>&nbsp;<tt></tt>
<p><tt>Rick Workman</tt>
<br><tt>Nortel Networks</tt>
<br><tt>Ottawa</tt>
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<br><tt>In an attempt to get some closure on the SIP-URL header BNF, how
about:</tt>
<p><tt>headers&nbsp;&nbsp;&nbsp;&nbsp; = "?" header *("&amp;" header)</tt>
<br><tt>header&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = hname "=" hvalue</tt>
<br><tt>hname&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = 1*(hname_char)</tt>
<br><tt>hvalue&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = *(hvalue_char)</tt>
<p><tt>hname_char&nbsp; = alphanum | "-" | "." | "!" | "*" | "_" | "+"
| "'" | "~"</tt>
<p><tt>hvalue_char = hvreserved | unreserved | escaped</tt>
<br><tt>hvreserved&nbsp; = ";" | "/" | "?" | ":" | "@" | "=" | "+" |"$"</tt>
<br>&nbsp;
<p><tt>Notes:</tt>
<br><tt>1. This isn't strictly backward compatible but I can't see any
practical issues. (Same as proposed change to other-param?)</tt>
<br><tt>2. hname_char set is a subset of token (to be consistent with "field-name")
which doesn't include "%" (URI "delims") or "`" (URI "unwise").</tt>
<br><tt>3. hvreserved is a subset of the current reserved rule eliminating
"&amp;", "," as per earlier discussion, and "[" and "]" (both URI "unwise").</tt>
<br><tt>4. Scanning the current header definitions, at least the following
must be escaped:</tt>
<br><tt>&nbsp;&nbsp; - any required LWS, e.g., for token separation or
in rfc1123 dates</tt>
<br><tt>&nbsp;&nbsp; - double quotes in quoted strings</tt>
<br><tt>&nbsp;&nbsp; - "\" in quoted-pair (comments and quoted-strings)</tt>
<br><tt>&nbsp;&nbsp; - "%" and "`" in tokens</tt>
<br><tt>&nbsp;&nbsp; - "&lt;" and ">" in name-addrs</tt>
<br><tt>&nbsp;&nbsp; - "&amp;" in any embedded SIP-URL headers</tt>
<br><tt>&nbsp;&nbsp; - "," in # lists</tt>
<br><tt>&nbsp;&nbsp; - any illegal characters in TEXT-UTF8</tt>
<br><tt>&nbsp;&nbsp; - any others?</tt>
<br><tt>5. Multi-byte UTF-8 characters will be represent by a 2*escaped
, i.e., the UTF-8 character CA98 will escape to %CA%98. (I think this was
the intent of Jonathan Lennox's previous proposal.)</tt>
<br>&nbsp;
<p><tt>Also, I'd like to see the text concerning reserved characters and
escaping clarified. If the BNF can be tightened up to the point where the
allowable characters for any SIP-URL component have been clearly defined,
then, by definition, any other characters must be escaped. Calling these
other characters reserved confuses them with those in the "reserved" rule
in the URI spec (and the current SIP RFC). Am I missing something?</tt>
<br>&nbsp;
<p><tt>Rick Workman</tt>
<br><tt>Nortel Networks</tt>
<br><tt>Ottawa</tt></blockquote>
</html>

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Hello,

RFC 2543 specifies warning header as optional in the INVITE method. But
the chapter describing Warning (6.41 Warning p 72) only details the
usage of that header in case of a response. So my question is: what is
the purpose of the warning header in an INVITE message ?

Best regards,

--
Francois-Xavier Guitton (SWD/ULC)
mailto:Francois-Xavier.Guitton@ms.alcatel.fr
Tel: +33 (0)1 69 63 47 52
Fax: +33 (0)1 69 63 17 89



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Hello,
<p>RFC 2543 specifies warning header as optional in the INVITE&nbsp;method.
But the chapter describing Warning (6.41 Warning p 72) only details the
usage of that header in case of a response. So my question is: what is
the purpose of the warning header in an INVITE&nbsp;message ?
<p>Best regards,
<pre>--&nbsp;
Francois-Xavier Guitton (SWD/ULC)
<A HREF="mailto:Francois-Xavier.Guitton@ms.alcatel.fr">mailto:Francois-Xavier.Guitton@ms.alcatel.fr</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Tel: +33 (0)1 69 63 47 52&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Fax: +33 (0)1 69 63 17 89</pre>
&nbsp;</html>

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> Hello,
> 
> RFC 2543 specifies warning header as optional in the INVITE method. But
> the chapter describing Warning (6.41 Warning p 72) only details the
> usage of that header in case of a response. So my question is: what is
> the purpose of the warning header in an INVITE message ?
> 
> Best regards,
> 
> --
> Francois-Xavier Guitton (SWD/ULC)
> mailto:Francois-Xavier.Guitton@ms.alcatel.fr
> Tel: +33 (0)1 69 63 47 52
> Fax: +33 (0)1 69 63 17 89

I was unable to find any place in the bis draft that stated Warning: was optional
for the INVITE request. The table on page 36, section 6.1 indicates that the
Warning: header is optional for INVITE responses (the lowercase r in the "where" 
column).

--
Sean Olson <sean.olson@ericsson.com>
Ericsson Inc.

 


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Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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James Kempf wrote:
> 
> Couple of comments.
> 
> >3. AUTHENTICATION (end-to-end)
> >==============================
> >We have the following potential authentication requirements:
> >- Called party id
> >- Calling party id
> >- Who we thought we were calling
> >
> 
> This doesn't say anything about a client authenticating a proxy,
> but that seems to be implied by the subsequent discussion.

e2e authentication is not about client authenticating a proxy, its about
clients authenticating clients. hBh authentication involves client
authentication of proxies and vice a versa.

> 
> >Dean Willis and Robert Sparks objected that you need app-level security
> >when you don’t
> >have transitivity of trust, and there are applications where you do not
> >have transitivity
> >of trust. For example, consider a call where you involve multiple
> >providers for that call
> >(multi-proxy authenticate). Basically, it’s not enough to have either
> >end-to-end or
> >hop-by-hop security; you also need to have end-to-intermediate security.
> >
> >This was a different model than had been worked on for the majority of
> >the meeting
> >however no one argued against the validity of it. Since it contradicts
> >some of the
> >earlier assumptions and decisions made during the meeting, each of these
> >will have to be
> >revisited.
> >
> 
> I`m not exactly sure what is meant by "application level security" here.
> Does "application" mean "application built on top of SIP" or does
> it mean "SIP as an application protocol?

The latter.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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James Kempf writes:
 > I`m not exactly sure what is meant by "application level security" here.
 > Does "application" mean "application built on top of SIP" or does
 > it mean "SIP as an application protocol?

   The latter.

 > Certainly in the case of cellular, it will be the case that the
 > proxy will be changing. When the mobile terminal roams outside
 > of it's cellular provider's domain, the new proxy needs to be authenticated
 > with the mobile terminal, and vice versa. 

   That sort of pre-supposes that the proxy the
   handset contacts when it's roamed isn't its
   home proxy. I'm not sure I see why that
   wouldn't still be the case. I hope people
   aren't getting buffaloed into some garden
   walls kind of mentality where the service
   provider gets to keep their stranglehold on
   services.

   If you ask me, the only thing I want the
   cellular service provider to provide me is
   bits on the air (ie, access, QoS). Where I get
   routing information, etc, should be my choice.
   Settling the bits on the air could have a 
   component which involves SIP (ala DQoS), but
   it shouldn't force me to get services through
   SlamAir if don't want to.

   Not that that has anything to do with SIP
   security.

		Mike


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> A number of people participated in the SIP Security Task Force meeting

Must have been a really secure meeting if the identity of the
participants is secret :-)

> Jonathan Rosenberg opened the discussion by suggesting that:
> - SIP encryption (not authentication) should be deprecated, because:

I strongly disagree with this. SIP encryption serves a number of useful
purposes. It hides information, for example, about Contact, Subject,
Organization, Retry-After, media types, media IP addresses and meeting
times (in SDP), SDP k= keys, etc. All of these are irrelevant for proxy
routing behavior, but highly sensitive. 

>   o It doesn’t work with NAT and firewalls

That's a risk you take. I refuse to dumb down protocols for everybody
just because some people are behind firewalls and NATs. There are lots
of protocols or features that you can't use behind a firewall or NAT.

>   o It assumes you know the public key of the party you are calling,
> which is not a good
> assumption.

It works in many cases. In other cases it can be made to work, e.g., via
indirection.

>   o It only really hides the body (and again that doesn’t work well due
> to the above).

See above.

> 
> Initial discussion of these points led to the decision that it should
> still be possible
> to support end-to-end encryption as there may be some uses for it.

Agreed.

> 
> A problem with the current end-to-end encryption scheme is, that it
> relies on knowing the
> public key of the other party. One way of solving that is to return a
> public key in the
> response and simply do a re-INVITE then. However, it was agreed that we
> should not do
> that, as we effectively end up duplicating IKE, or something similar.
> 
> Dave Oran suggested that end-to-end information that needs encryption
> should simply go in
> the body (don’t encrypt headers). We could use multipart for this, or
> simply allow for a
> single body to be encrypted and define an error code to be returned if
> the proxy needs to
> see the body (e.g. if SDP is encrypted and message traverses a
> firewall). The group
> decided to accept this mechanism. The error code will only support
> coarse granularity,
> i.e., if multiple bodies are encrypted, you will not be told which one
> is needed in
> clear, just a general error saying “body not understood”.

I had originally (many years ago) suggested a similar mechanism,
assuming I understand your proposal correctly, namely to have a body
type "SIP message", which is encrypted. A decision was made not to do
this (I'm afraid I no longer remember the reason; might have been
message size concerns or processing logic). If you only want to encrypt
SDP, I disagree, for the reasons stated above. If you want to do both
and use the "SIP message" body mechanism, I'm not against this, but this
does entail a significant protocol change, without changing any of the
other problems.

> 
> Additional discussion on encryption of headers ensued. Contact is an
> example of header
> that can be usefully encrypted (end-to-end), but in general, there
> doesn’t seem to be a
> whole lot of use for this (generally, if the information is relevant it
> can’t be
> encrypted). Only want the registrar to see the Contact, not any of the
> intermediate
> proxies.

Again, I disagree, see above.

> 3. AUTHENTICATION (end-to-end)
> ==============================

> 3.1 Hop-by-hop Authentication
> =============================

> We want to have at least one mechanism specified in the SIP
> specification.

Why? This gets into endless arguments as to which is better.

> 
> It was decided that we should have the following hop-by-hop security
> mechanisms:
> - Server to server: IPSec using ESP transport,
> - Client to server: IPSec using ESP transport (not clear ESP transport
> was agreed to)

Disagree. For practical reasons, TLS is much easier to deploy (keying,
available libraries, available infrastructure to obtain keys, etc.). Are
there any technical reasons to force somebody to implement IPsec?

> 
> 4. Miscellaneous
> ================
> 
> 4.1 Areas needing further work
> ==============================
> - Challenge-response doesn’t work with forking (but addresses replay) as
> only one 401
> will get back to the caller.

This can be solved, at the cost of increased proxy complexity, by having
the proxy gather all challenges from all branches and return them in the
401 response, similar to multi-proxy authentication.




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> 
> I believe what URI really means in the SIP context is absoluteURI. That
> is, there must always be a scheme, followed by a colon. I believe this
> eliminates many of the ambiguities people have been raising. I'm not a
> big fan of relative URIs within HTTP itself (as opposed to within HTML,
> where the browser computes the absolute URI and puts that in the HTTP
> request); in general, I believe URIs sent on the wire should be globally
> resolvable. HTTP ran into all sorts of problems, as I recall, since URIs
> were not always absolute, and thus servers that supported many domains
> got confused when they received things like GET index.html HTTP/1.0.

There are two kinds of relative URLs: omitting the host part (which is
what got HTTP/1.0 into trouble) and relative URLs that are relative to
the current file system location (as in ../../foo.html).  The first is
not helpful, the second not applicable.




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Henning Schulzrinne writes:
 > >   o It assumes you know the public key of the party you are calling,
 > > which is not a good
 > > assumption.
 > 
 > It works in many cases. In other cases it can be made to work, e.g., via
 > indirection.

   Just as a general note -- again -- any
   dependence on public key signatures/encryption
   on a per message basis will not scale. They
   may work in toy systems, but when you start
   talking about systems which must do 10-100's
   of transactions per second, they simply do 
   not scale: it is simply beyond the state of
   the art by several orders of magnitude.

   I have no particular qualms about having the
   ability to use public key operations as a
   quick and dirty (and stateless) way to do
   crypto, but we *must* have the ability to
   take anything which used a public key operation
   before and have a symmetric key analog for it,
   and SIP's must support it as a basic mechanism.   

   Obviously, symmetric key operation begs the 
   question of key distribution, of which there
   may be many choices -- I personally think that
   kerberos covers the most bases in a scalable
   fashion, though not all. However, even if the
   the key distribution is done by Palantir, we
   have to have a way to do this.

   I'm somewhat ignorant of PGP, so forgive me
   if this already part of it. All public key
   signatures and encryptions still rely on 
   symmetric key operations: signatures do a
   SHA-1 HMAC over the message and then encrypts
   the resulting hash with their private key. Encryption
   cons's up a key and runs 3DES over the message.
   It then encrypts the key itself with the
   sendee's public key. 

   In both cases, there is a base level symmetric
   key operation which is common. This should be
   abstracted such that either symmetric key or
   public key operations are a choice which is
   discernible at the receiver. As I read 2543
   right now, however, there seems to be a base
   level assumption that it is *always* done
   through public key.  If PGP doesn't allow 
   for symmetric key only, then the entire 
   mechanism is incomplete and needs to be 
   fixed. Even if PGP doesn't make that 
   assumption, it would be very nice in 
   2543bis to actually show that you don't need
   to always use public key operations as the
   current text is highly misleading.

 > > We want to have at least one mechanism specified in the SIP
 > > specification.
 > 
 > Why? This gets into endless arguments as to which is better.

   There were no endless arguments in MEGACO about
   this, which argued about everything endlessly.

 > > It was decided that we should have the following hop-by-hop security
 > > mechanisms:
 > > - Server to server: IPSec using ESP transport,
 > > - Client to server: IPSec using ESP transport (not clear ESP transport
 > > was agreed to)
 > 
 > Disagree. For practical reasons, TLS is much easier to deploy (keying,
 > available libraries, available infrastructure to obtain keys, etc.). Are
 > there any technical reasons to force somebody to implement IPsec?

   Yes: there is no such thing as TLS over UDP.
   Even if there is some experimental UDP based
   version, it negates the advantages you list
   above. Also: which OS's don't have IPsec these
   days? 

   Beyond that, TLS is the worst of all possible
   worlds: if I'm a programmer, I either want to
   have fine control on what portions of the
   stream I diddle at the added complexity of 
   understanding how the API and functionality
   works, or I want to be completely ignorant of
   what goes on at a lower layer. 

   TLS, however, requires that I learn the API
   *and* gives me none of the benefits of fine
   control. IPsec at least has the know-nothing
   property; the application can remain utterly
   ignorant. Considering the pathetic state of
   security right now, anything that puts a 
   requirement for a program to touch *anything*
   is tantamount to saying that it won't happen
   any time soon. IPsec at least gives users
   a third party frob to compensate for
   application inadequacy; TLS does not.

	       Mike


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Michael Thomas wrote:
> 
> Henning Schulzrinne writes:
>  > >   o It assumes you know the public key of the party you are calling,
>  > > which is not a good
>  > > assumption.
>  >
>  > It works in many cases. In other cases it can be made to work, e.g., via
>  > indirection.
> 
>    Just as a general note -- again -- any
>    dependence on public key signatures/encryption
>    on a per message basis will not scale. They
>    may work in toy systems, but when you start

They work just fine (modulo the PKI problem) in anything *but*
large-scale PSTN gateways. Presumably, gateways are a transient (albeit
important) problem.

>    talking about systems which must do 10-100's
>    of transactions per second, they simply do
>    not scale: it is simply beyond the state of
>    the art by several orders of magnitude.
> 
>    I have no particular qualms about having the
>    ability to use public key operations as a
>    quick and dirty (and stateless) way to do
>    crypto, but we *must* have the ability to
>    take anything which used a public key operation
>    before and have a symmetric key analog for it,
>    and SIP's must support it as a basic mechanism.
> 
>    Obviously, symmetric key operation begs the
>    question of key distribution, of which there
>    may be many choices -- I personally think that
>    kerberos covers the most bases in a scalable
>    fashion, though not all. However, even if the
>    the key distribution is done by Palantir, we
>    have to have a way to do this.

I have yet to see a scalable, inter-domain Kerberos system. What makes
you believe that this is easier than regular public key?

Most intelligent-endsystem (as opposed to PSTN gateway) calls will be
essentially random and sparse in time, with intervals between calling
the same system longer than the ticket lifetime, so that rekeying will
be needed for each call. I can't imagine that this scales any better.

> 
>    I'm somewhat ignorant of PGP, so forgive me
>    if this already part of it. All public key
>    signatures and encryptions still rely on
>    symmetric key operations: signatures do a
>    SHA-1 HMAC over the message and then encrypts
>    the resulting hash with their private key. Encryption
>    cons's up a key and runs 3DES over the message.
>    It then encrypts the key itself with the
>    sendee's public key.
> 
>    In both cases, there is a base level symmetric
>    key operation which is common. This should be
>    abstracted such that either symmetric key or
>    public key operations are a choice which is
>    discernible at the receiver. As I read 2543
>    right now, however, there seems to be a base
>    level assumption that it is *always* done
>    through public key.  If PGP doesn't allow
>    for symmetric key only, then the entire
>    mechanism is incomplete and needs to be
>    fixed. Even if PGP doesn't make that
>    assumption, it would be very nice in
>    2543bis to actually show that you don't need
>    to always use public key operations as the
>    current text is highly misleading.

Again, except in the case of high-volume signaling trunks between
gateways, interarrival times for calls between the same
source-destination pair may well exceed customary ticket lifetimes. If
we do have high-volume signaling trunks, we have existing resaonably
efficinet mechanisms (IPsec, TLS) to handle this. I'm really reluctant
to cook up a new mechanism until we know that the existing ones cannot
do the trick.

>    There were no endless arguments in MEGACO about
>    this, which argued about everything endlessly.

What's your concrete proposal? We'll see what arguments we get (or
not...).

>    Yes: there is no such thing as TLS over UDP.

For high-volume signaling trunks where key exchanges and operations
matters, TCP is probably the more appropriate solution.

>    Even if there is some experimental UDP based
>    version, it negates the advantages you list
>    above. Also: which OS's don't have IPsec these
>    days?

Windows 98 and Windows NT come to mind, at least out of the box. From
all that I heard, most of the IPsec versions do not interoperate across
OS or implementations. I believe there was even a report on installing
IPsec on this list which struck me as basically requiring deep IPsec
understanding to make anything work.

Again: I have no problem with IPsec. I do have a problem with not
supporting TLS.

> 
>    Beyond that, TLS is the worst of all possible
>    worlds: if I'm a programmer, I either want to
>    have fine control on what portions of the
>    stream I diddle at the added complexity of
>    understanding how the API and functionality
>    works, or I want to be completely ignorant of
>    what goes on at a lower layer.
> 
>    TLS, however, requires that I learn the API
>    *and* gives me none of the benefits of fine
>    control. IPsec at least has the know-nothing
>    property; the application can remain utterly
>    ignorant. Considering the pathetic state of
>    security right now, anything that puts a
>    requirement for a program to touch *anything*
>    is tantamount to saying that it won't happen
>    any time soon. IPsec at least gives users
>    a third party frob to compensate for
>    application inadequacy; TLS does not.

How come that millions of web servers and other applications actually
use TLS, on a daily basis?




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From: James Kempf <James.Kempf@Eng.Sun.COM>
Reply-To: James Kempf <James.Kempf@Eng.Sun.COM>
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
To: James.Kempf@Eng.Sun.COM, mat@cisco.com
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> > Certainly in the case of cellular, it will be the case that the
> > proxy will be changing. When the mobile terminal roams outside
> > of it's cellular provider's domain, the new proxy needs to be authenticated
> > with the mobile terminal, and vice versa. 
>
>   That sort of pre-supposes that the proxy the
>   handset contacts when it's roamed isn't its
>   home proxy. I'm not sure I see why that
>   wouldn't still be the case. I hope people
>   aren't getting buffaloed into some garden
>   walls kind of mentality where the service
>   provider gets to keep their stranglehold on
>   services.
>

It's not the case because if I have my cellular account with a provider
in California and I travel to Australia, I don't want to have to 
go all the way back to the SIP proxy with my provider in California
to have to make a call.


		jak



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From: Michael Thomas <mat@cisco.com>
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To: Henning Schulzrinne <hgs@cs.columbia.edu>
Cc: Michael Thomas <mat@cisco.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        sip <sip@lists.bell-labs.com>
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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Henning Schulzrinne writes:
 > Michael Thomas wrote:
 > >    Just as a general note -- again -- any
 > >    dependence on public key signatures/encryption
 > >    on a per message basis will not scale. They
 > >    may work in toy systems, but when you start
 > 
 > They work just fine (modulo the PKI problem) in anything *but*
 > large-scale PSTN gateways. Presumably, gateways are a transient (albeit
 > important) problem.

   Um, no. Proxies which need to scale to 10's
   or 100's of thousands of users have the same
   problem. With current state of art, you could
   not build a proxy that big which requires public
   key operations. Low end gateways (25Mhz class) 
   take hundreds of milliseconds -- I've heard
   going on a second -- to do one public operation.
   That is directly in the path of post dial
   delay.

   I don't see how you can handwave those issues
   away as unimportant. Large scale SIP
   deployments are hopefully going to do better
   than the PSTN -- and the rest of the Internet
   -- on security, but it certainly will not and
   can not happen if the bar is set too high.

   Also: it's telling that you brush the PKI
   problem aside as a detail. There's a reason
   why there's no such thing as a large scale
   PKI after 20 years of having the base technology: 
   they're incredibly hard -- maybe insurmountably
   hard -- things to build.

 > I have yet to see a scalable, inter-domain Kerberos system. What makes
 > you believe that this is easier than regular public key?

   Well, for one it allows you to vastly scale
   down the reliance on public key infrastructure
   to a set of potentially high-touch machines
   (the KDC). Are you familiar with PKCROSS?

   Besides, you can do quite a lot with Kerberos
   without ever even getting into the cross realm
   tar pit. 

 > Most intelligent-endsystem (as opposed to PSTN gateway) calls will be
 > essentially random and sparse in time, with intervals between calling
 > the same system longer than the ticket lifetime, so that rekeying will
 > be needed for each call. 

   Well, there are more things than PSTN gateways
   which have scaling problems. IVR maze machines,
   conference bridges, proxies, etc, etc, will
   have identical scaling concerns. Kerberos
   tickets can also have a long lifetime --
   days, weeks even. While it's questionable
   whether my IP phone at home would want to
   to have a service ticket for the end to end
   call, it's unquestionable that it would be
   a benefit to use Kerberos to key up the SA
   between the UA and its first hop proxy. *One*
   *call* would be enough to amortize the cost.
 
>  I can't imagine that this scales any better.

   The only thing you can say is that Kerberos
   degenerates into public key case in
   pathological situations. That's not a very
   strong starting point.

 > Again, except in the case of high-volume signaling trunks between
 > gateways, interarrival times for calls between the same
 > source-destination pair may well exceed customary ticket lifetimes. 

   End to end is not the only consideration. In
   fact, it's the least interesting because I 
   might just as well ship the end to end SIP keying
   material through in the initial INVITE as plain
   text if the destination is in the same trust
   domain as the KDC. 

 >If
 > we do have high-volume signaling trunks, we have existing resaonably
 > efficinet mechanisms (IPsec, TLS) to handle this. 

   I hope that you're not thinking that IPsec with
   IKE to do end to end keying would be viable
   between trunking gateways and high fan out
   UA's. TLS would be no better.

 > I'm really reluctant
 > to cook up a new mechanism until we know that the existing ones cannot
 > do the trick.

   Kerberos is a new mechanism? I guess if you
   consider the late 80's "new". SIP, and maybe
   HTTP, are conspicuous in their lack of a
   Kerberos mechanism. You can even use Kerberos
   to key RSVP integrity objects.

   And I don't know how much more you need to
   "know" here: I work for a manufacturer of
   a lot of the components that would be needed
   in a large scale SIP system. I know that it
   cannot be built now as specified, and I'm
   doubtful that it can be built any time soon. I
   do know, however, that a relatively simple and
   well understood piece of technology can make the
   entire problem moot. It seems like a no-brainer
   to me.

   Also as a point of reference: I think that
   SIP should recommend IPsec as the base hop
   by hop security mechanism. I don't think it
   needs to say anything about which key
   distribution protocol is recommended though.
   I have my opinions, but the spec doesn't need
   to reflect them.

 > What's your concrete proposal? We'll see what arguments we get (or
 > not...).

   The full protocol is being worked on as we
   speak. You can also look at the security spec
   on packetcable.com to get an idea of how you
   can do Kerberized IPsec for SIP. It's currently
   spec'd for MGCP, but the transported protocol
   is actually irrelevant.

 > >    Yes: there is no such thing as TLS over UDP.
 > 
 > For high-volume signaling trunks where key exchanges and operations
 > matters, TCP is probably the more appropriate solution.

   That's highly arguable; I'm sure the SCTP folks
   might have thought or two about that, and as
   far as I can tell, they have the full support of
   their AD's. 
 
 > Windows 98 and Windows NT come to mind, at
 > least out of the box. 

   Win2k and presumably (?) that other Windows
   market churner will have it. And you can always
   do bump in the stack and bump in the wire
   implementations of IPsec.

 > From
 > all that I heard, most of the IPsec versions do not interoperate across
 > OS or implementations.

   The problem isn't IPsec usually, the problem is 
   with IKE. 

 > Again: I have no problem with IPsec. I do have a problem with not
 > supporting TLS.

   But TLS doesn't address one quite legitimate
   and useful operational mode of SIP: UDP with
   ALF. That's quite a shortcoming. If we
   recommend any hop by hop security, it at the
   very least needs to work with all operational
   modes of SIP.

	    Mike


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To: James Kempf <James.Kempf@Eng.Sun.COM>
Cc: mat@cisco.com, sip@lists.bell-labs.com, jdrosen@dynamicsoft.com
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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James Kempf writes:
 > >   That sort of pre-supposes that the proxy the
 > >   handset contacts when it's roamed isn't its
 > >   home proxy. I'm not sure I see why that
 > >   wouldn't still be the case. I hope people
 > >   aren't getting buffaloed into some garden
 > >   walls kind of mentality where the service
 > >   provider gets to keep their stranglehold on
 > >   services.
 > >
 > 
 > It's not the case because if I have my cellular account with a provider
 > in California and I travel to Australia, I don't want to have to 
 > go all the way back to the SIP proxy with my provider in California
 > to have to make a call.

   Why? Seriously. SIP is not a soft realtime protocol
   like, say, MGCP where you need tight timing
   tolerances. I can think of a myriad of reasons
   why my SIP cell phone doesn't want the local
   provider to be anything but a bits in the air
   transporter. I trust my SIP services to the
   service provider with whom I contract; I don't
   don't want slammer.au to have anything to do 
   with delivering my services other than what
   base level physics requires.

	      Mike


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From: "Paul E. Jones" <paul.jones@ties.itu.ch>
To: <S_S_Dinesh/India/IBM@au1.ibm.com>, <sip@lists.research.bell-labs.com>,
        <iptel@lists.research.bell-labs.com>
References: <CA2568D9.0036C741.00@d73mta05.au.ibm.com>
Date: Mon, 15 May 2000 05:35:50 -0400
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Subject: [SIP] Re: [IPTEL] RE: TRIP for gateway registrations
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While it is true that RAS is not used to register a large number of =
addresses, that will not be true in the future.  H.323v4 supports two =
mechanisms to allow MCUs or GWs to register large numbers of addresses =
and address ranges.

Paul
  ----- Original Message -----=20
  From: S_S_Dinesh/India/IBM@au1.ibm.com=20
  To: sip@lists.research.bell-labs.com ; =
iptel@lists.research.bell-labs.com=20
  Sent: Monday, May 08, 2000 4:40 AM
  Subject: [IPTEL] RE: TRIP for gateway registrations




  I would point out that H.323 RAS messages are not normally
  used to register a range of phone numbers for gateways as it
  is quite impractical, especially for gateways handling a large
  number of phone numbers, and when keep-alive messages are used.

  While some implementations uses tricks such as registering
  "dialling prefixes" to map to telephone number, it is not the
  intent of prefixes which are supposed to be used for special services.

  Normally, a gateway registers an alias with its gatekeeper and the =
binding
  between the alias and the range of phone numbers handled by the
  gateway is "manually" configured in the gatekeeper, as described
  in the draft. I would think that TRIP could automate this process.

  > -----Original Message-----
  > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
  > Sent: Thursday, March 09, 2000 11:33 PM
  > To: sip@lists.research.bell-labs.com;
  > iptel@lists.research.bell-labs.com
  > Subject: TRIP for gateway registrations
  >
  >
  > Folks,
  >
  > Every once in a while, the question comes up of why SIP
  > REGISTER is not
  > the right vehicle for allowing gateways to register phone
  > prefixes with
  > a proxy. I believe TRIP is far better for this purpose. As it
  > turns out,
  > a small subset of TRIP is needed to solve this problem, and yet its
  > still interoperable with full blown TRIP. Hussein Salama and I have
  > written an I-D that discusses this. Until it appears in the =
archives,
  > you can pick up a copy at:
  >
  > http://www.cs.columbia.edu/~jdrosen/papers/draft-rs-trip-gw-00.txt
  >
  > Thanks,
  > Jonathan R.
  > --
  > Jonathan D. Rosenberg                       200 Executive Drive
  > Chief Scientist                             Suite 120
  > dynamicsoft                                 West Orange, NJ 07052
  > jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
  > http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
  > http://www.dynamicsoft.com
  >
  >


------=_NextPart_000_005C_01BFBE2F.6DC04240
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	charset="Windows-1252"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Dwindows-1252" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.3013.2600" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial>While it is true that RAS is not used to =
register a large=20
number of addresses, that will not be true in the future.&nbsp; H.323v4 =
supports=20
two mechanisms to allow MCUs or GWs to register large numbers of =
addresses and=20
address ranges.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial>Paul</FONT></DIV>
<BLOCKQUOTE=20
style=3D"BORDER-LEFT: #000000 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: =
0px; PADDING-LEFT: 5px; PADDING-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 href=3D"mailto:S_S_Dinesh/India/IBM@au1.ibm.com"=20
  =
title=3DS_S_Dinesh/India/IBM@au1.ibm.com>S_S_Dinesh/India/IBM@au1.ibm.com=
</A>=20
  </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A=20
  href=3D"mailto:sip@lists.research.bell-labs.com"=20
  =
title=3Dsip@lists.research.bell-labs.com>sip@lists.research.bell-labs.com=
</A> ;=20
  <A href=3D"mailto:iptel@lists.research.bell-labs.com"=20
  =
title=3Diptel@lists.research.bell-labs.com>iptel@lists.research.bell-labs=
.com</A>=20
  </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Monday, May 08, 2000 4:40 =
AM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> [IPTEL] RE: TRIP for =
gateway=20
  registrations</DIV>
  <DIV><BR></DIV><BR><BR>I would point out that H.323 RAS messages are =
not=20
  normally<BR>used to register a range of phone numbers for gateways as =
it<BR>is=20
  quite impractical, especially for gateways handling a large<BR>number =
of phone=20
  numbers, and when keep-alive messages are used.<BR><BR>While some=20
  implementations uses tricks such as registering<BR>"dialling prefixes" =
to map=20
  to telephone number, it is not the<BR>intent of prefixes which are =
supposed to=20
  be used for special services.<BR><BR>Normally, a gateway registers an =
alias=20
  with its gatekeeper and the binding<BR>between the alias and the range =
of=20
  phone numbers handled by the<BR>gateway is "manually" configured in =
the=20
  gatekeeper, as described<BR>in the draft. I would think that TRIP =
could=20
  automate this process.<BR><BR>&gt; -----Original Message-----<BR>&gt; =
From:=20
  Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]<BR>&gt; Sent: =
Thursday,=20
  March 09, 2000 11:33 PM<BR>&gt; To: =
sip@lists.research.bell-labs.com;<BR>&gt;=20
  iptel@lists.research.bell-labs.com<BR>&gt; Subject: TRIP for gateway=20
  registrations<BR>&gt;<BR>&gt;<BR>&gt; Folks,<BR>&gt;<BR>&gt; Every =
once in a=20
  while, the question comes up of why SIP<BR>&gt; REGISTER is =
not<BR>&gt; the=20
  right vehicle for allowing gateways to register phone<BR>&gt; prefixes =

  with<BR>&gt; a proxy. I believe TRIP is far better for this purpose. =
As=20
  it<BR>&gt; turns out,<BR>&gt; a small subset of TRIP is needed to =
solve this=20
  problem, and yet its<BR>&gt; still interoperable with full blown TRIP. =
Hussein=20
  Salama and I have<BR>&gt; written an I-D that discusses this. Until it =
appears=20
  in the archives,<BR>&gt; you can pick up a copy at:<BR>&gt;<BR>&gt;=20
  =
http://www.cs.columbia.edu/~jdrosen/papers/draft-rs-trip-gw-00.txt<BR>&gt=
;<BR>&gt;=20
  Thanks,<BR>&gt; Jonathan R.<BR>&gt; --<BR>&gt; Jonathan D.=20
  =
Rosenberg&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  200 Executive Drive<BR>&gt; Chief=20
  =
Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  Suite 120<BR>&gt;=20
  =
dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  West Orange, NJ 07052<BR>&gt;=20
  =
jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  FAX:&nbsp;&nbsp; (732) 741-4778<BR>&gt;=20
  =
http://www.cs.columbia.edu/~jdrosen&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;=20
  PHONE: (732) 741-7244<BR>&gt;=20
http://www.dynamicsoft.com<BR>&gt;<BR>&gt;<BR></BLOCKQUOTE></BODY></HTML>=


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Ladies and Gentlemen!

Sorry in case you receive multiple copies of this emmail,
which means your are very active ;-)


Computer Networks Journal plans to publish in February 2001 a special
issue on "Voice / Data Integration - A Snapshot of Intelligent Network
and Interent Convergence".

Guest editors: T. Magedanz (IKV++) and M. Smirnov (GMD FOKUS)

Deadline for submission is June 5th, 2000!!!

Maybe a good candidate for disseminating your results!

For more infos look at: http://www.fokus.gmd.de/events/IN-Internet/

Best regards,
Thomas Magedanz (magedanz@ikv.de)

IKV++ GmbH, Bernburger Str. 24-25, D-10963 Berlin, Germany




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From: "Michael Welser" <michael.welser@aon.at>
To: "sip" <sip@lists.bell-labs.com>
Subject: RE: [SIP] Minutes of SIP Security task force in Adelaide
Date: Mon, 15 May 2000 14:24:13 +0200
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> >   o It doesn’t work with NAT and firewalls
>
> That's a risk you take. I refuse to dumb down protocols for everybody
> just because some people are behind firewalls and NATs. There are lots
> of protocols or features that you can't use behind a firewall or NAT.

This opinion (attitude?) leads eventually to the discussions last heard
in foglamps.

I do not like NATs. But NATs are here, and they are reality for more
than just "some people" (most of which do not even have a say about
this) and they will be with us for quite some while.

I refuse to smarten up protocols for everybody (thereby cutting off
potential users from potential services) just because some people are
NOT behind firewalls and NATs. There are some protocols or features that
you can use behind a firewall or a NAT (otherwise the Internet wouldn't
be what it is now).

Michael Welser

------------------------------------------------------------
mailto:mwelser@highway.telekom.at         Telekom Austria AG
tel: +43 (1) 79711 1622                               Abt RS
fax: +43 (1) 79711 1608                    Arsenal Objekt 22
http://www.telekom.at                           Postfach 111
http://www.aon.at                                A-1103 Wien




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Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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Michael Thomas wrote:

>  > Certainly in the case of cellular, it will be the case that the
>  > proxy will be changing. When the mobile terminal roams outside
>  > of it's cellular provider's domain, the new proxy needs to be authenticated
>  > with the mobile terminal, and vice versa.
>
>    That sort of pre-supposes that the proxy the
>    handset contacts when it's roamed isn't its
>    home proxy. I'm not sure I see why that
>    wouldn't still be the case.

the current assumption is to have a "proxy" in the visited domain (it might be
something more than a pure proxy).

There are many reasons behind this choice (I might even be unaware of many
others...). Just three examples:

   * possible optimizations (e.g.: signalling for a basic call to an Australian
     fixed network number from an Italian mobile roaming to Australia could be
     processed locally, without any "tromboning")
   * some people claim the "proxy" should also perform some kind of admission
     control for the calls. The proxy in the home domain has no information on the
     visited domain
   * the PNO who runs the visited network might desire to have some kind of
     knowledge on what is going on in his network, from a call control point of
     view

Giuseppe

----------------------------------------------------------
Giuseppe RICAGNI
System Architecture and Product Planning
Broadband Networks
Italtel
via Reiss Romoli - C10
20019 Castelletto di Settimo Milanese (MILANO)
ITALY

mailto:giuseppe.ricagni@italtel.it
----------------------------------------------------------


--------------55383782A551F64459ED073B
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Michael Thomas wrote:
<blockquote TYPE=CITE>&nbsp;> Certainly in the case of cellular, it will
be the case that the
<br>&nbsp;> proxy will be changing. When the mobile terminal roams outside
<br>&nbsp;> of it's cellular provider's domain, the new proxy needs to
be authenticated
<br>&nbsp;> with the mobile terminal, and vice versa.
<p>&nbsp;&nbsp; That sort of pre-supposes that the proxy the
<br>&nbsp;&nbsp; handset contacts when it's roamed isn't its
<br>&nbsp;&nbsp; home proxy. I'm not sure I see why that
<br>&nbsp;&nbsp; wouldn't still be the case.</blockquote>
the current assumption is to have a "proxy" in the visited domain (it might
be something more than a pure proxy).
<p>There are many reasons behind this choice (I might even be unaware of
many others...). Just three examples:
<ul>
<li>
possible optimizations (e.g.: signalling for a basic call to an Australian
fixed network number from an Italian mobile roaming to Australia could
be processed locally, without any "tromboning")</li>

<li>
some people claim the "proxy" should also perform some kind of admission
control for the calls. The proxy in the home domain has no information
on the visited domain</li>

<li>
the PNO who runs the visited network might desire to have some kind of
knowledge on what is going on in his network, from a call control point
of view</li>
</ul>
Giuseppe
<p>----------------------------------------------------------
<br>Giuseppe RICAGNI
<br>System Architecture and Product Planning
<br>Broadband Networks
<br>Italtel
<br>via Reiss Romoli - C10
<br>20019 Castelletto di Settimo Milanese (MILANO)
<br>ITALY
<p><A HREF="mailto:giuseppe.ricagni@italtel.it">mailto:giuseppe.ricagni@italtel.it</A>
<br>----------------------------------------------------------
<br>&nbsp;</html>

--------------55383782A551F64459ED073B--




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From sip-admin@lists.bell-labs.com  Mon May 15 13:44:28 2000
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From: "Robert Johnson" <rjohnson@adventnetworks.com>
To: <welser@acm.org>, "'sip'" <sip@lists.bell-labs.com>
Subject: RE: [SIP] Minutes of SIP Security task force in Adelaide
Date: Mon, 15 May 2000 12:42:18 -0500
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>> >   o It doesn’t work with NAT and firewalls
>>
>> That's a risk you take. I refuse to dumb down protocols for everybody
>> just because some people are behind firewalls and NATs. There are lots
>> of protocols or features that you can't use behind a firewall or NAT.
>
>This opinion (attitude?) leads eventually to the discussions last heard
>in foglamps.
>
>I do not like NATs. But NATs are here, and they are reality for more
>than just "some people" (most of which do not even have a say about
>this) and they will be with us for quite some while.
>
>I refuse to smarten up protocols for everybody (thereby cutting off
>potential users from potential services) just because some people are
>NOT behind firewalls and NATs. There are some protocols or features that
>you can use behind a firewall or a NAT (otherwise the Internet wouldn't
>be what it is now).

If we do not interwork with NATs, SIP will not fly. Most commercial Internet
subscribers today pass through a NAT, whether they know it or not. However,
deprecating SIP encryption forces implementers into incompatible product and
network configurations, which is also a limiter to the growth of SIP. Like most
enterprise H.323 implementations, most SIP implementations will operate
completely within the enterprise domain, going outside through a proxy that is
positioned to avoid the NAT/FW issue entirely. This, along with other
arguments, leads me to think that deprecating SIP encryption is not necessarily
the wisest move.

Robert Johnson
Director of Network Architecture
Advent Networks
(512) 397-4914

SIP mailing list
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        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: RE: [SIP] Minutes of SIP Security task force in Adelaide
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> -----Original Message-----
> From:	Henning Schulzrinne [SMTP:hgs@cs.columbia.edu]
> Sent:	Saturday, May 13, 2000 3:46 AM
> To:	Jonathan Rosenberg
> Cc:	sip
> Subject:	Re: [SIP] Minutes of SIP Security task force in Adelaide
> 
> > A number of people participated in the SIP Security Task Force meeting
> 
> Must have been a really secure meeting if the identity of the
> participants is secret :-)
> 
> > Jonathan Rosenberg opened the discussion by suggesting that:
> > - SIP encryption (not authentication) should be deprecated, because:
> 
> I strongly disagree with this. SIP encryption serves a number of useful
> purposes. It hides information, for example, about Contact, Subject,
> Organization, Retry-After, media types, media IP addresses and meeting
> times (in SDP), SDP k= keys, etc. All of these are irrelevant for proxy
> routing behavior, but highly sensitive. 
> 
	I agree with the disagreement. 

> >   o It doesn't work with NAT and firewalls
> 
> That's a risk you take. I refuse to dumb down protocols for everybody
> just because some people are behind firewalls and NATs. There are lots
> of protocols or features that you can't use behind a firewall or NAT.
> 
	One could consider a SIP proxy as an ALG and a firewall element,
could one not? If you agree with this then it does work for NAT and
firewalls. 

	I really sick of people using "NAT" as a logical fallacy in their
arguments. 

	Perhaps if we bury our heads in the sand long enough they will go
away. 

> >   o It assumes you know the public key of the party you are calling,
> > which is not a good
> > assumption.
> 
> It works in many cases. In other cases it can be made to work, e.g., via
> indirection.
> 
	Yep

> >   o It only really hides the body (and again that doesn't work well due
> > to the above).
> 
> See above.
> 
> > 
> > Initial discussion of these points led to the decision that it should
> > still be possible
> > to support end-to-end encryption as there may be some uses for it.
> 
> Agreed.
> 
> > 
> > A problem with the current end-to-end encryption scheme is, that it
> > relies on knowing the
> > public key of the other party. One way of solving that is to return a
> > public key in the
> > response and simply do a re-INVITE then. However, it was agreed that we
> > should not do
> > that, as we effectively end up duplicating IKE, or something similar.
> > 
> > Dave Oran suggested that end-to-end information that needs encryption
> > should simply go in
> > the body (don't encrypt headers). We could use multipart for this, or
> > simply allow for a
> > single body to be encrypted and define an error code to be returned if
> > the proxy needs to
> > see the body (e.g. if SDP is encrypted and message traverses a
> > firewall). The group
> > decided to accept this mechanism. The error code will only support
> > coarse granularity,
> > i.e., if multiple bodies are encrypted, you will not be told which one
> > is needed in
> > clear, just a general error saying "body not understood".
> 
> I had originally (many years ago) suggested a similar mechanism,
> assuming I understand your proposal correctly, namely to have a body
> type "SIP message", which is encrypted. A decision was made not to do
> this (I'm afraid I no longer remember the reason; might have been
> message size concerns or processing logic). If you only want to encrypt
> SDP, I disagree, for the reasons stated above. If you want to do both
> and use the "SIP message" body mechanism, I'm not against this, but this
> does entail a significant protocol change, without changing any of the
> other problems.
> 
> > 
> > Additional discussion on encryption of headers ensued. Contact is an
> > example of header
> > that can be usefully encrypted (end-to-end), but in general, there
> > doesn't seem to be a
> > whole lot of use for this (generally, if the information is relevant it
> > can't be
> > encrypted). Only want the registrar to see the Contact, not any of the
> > intermediate
> > proxies.
> 
> Again, I disagree, see above.
> 
	I see a plethora of valuable uses for separate encryption process
for the Contact and many other headers. Especially when there are
hierarchies or network ownership boundaries between UAs, proxies,
registrars, ALGs, etc..

> > 3. AUTHENTICATION (end-to-end)
> > ==============================
> 
> > 3.1 Hop-by-hop Authentication
> > =============================
> 
> > We want to have at least one mechanism specified in the SIP
> > specification.
> 
> Why? This gets into endless arguments as to which is better.
> 
	Industry standards consortia should pick one for the network they
are designing - not the protocol. If all of the public infrastructure
standards consortia agree to use the end to end SA model then that's great. 

	SIP is a protocol that should be flexible enough to support many
mechanisms. 

	For instance I might want to use one mechanism for an in-house
solution while the public infrastructure may warrant something else.

	Then again, a military or government organization may wish to use
something completely different.

	Then again, a service provider may way to do something extra if the
communications are on net.

> > 
> > It was decided that we should have the following hop-by-hop security
> > mechanisms:
> > - Server to server: IPSec using ESP transport,
> > - Client to server: IPSec using ESP transport (not clear ESP transport
> > was agreed to)
> 
> Disagree. For practical reasons, TLS is much easier to deploy (keying,
> available libraries, available infrastructure to obtain keys, etc.). Are
> there any technical reasons to force somebody to implement IPsec?
> 
	This is but one method. Go to a consortia group to get this
approved. If a vendor doesn't want to support it - then don't.

	I do agree that this is a good and scalable choice for a consortia
group building a public infrastructure network to make; however, I bet
you'll see the other stuff used as well if not in addition to. I guess what
I'm saying is "good start" but this choice is probably a bit too simplistic
and utopian.

	Many of these mechanisms are compatible.

	How many actual providers were at the meeting voicing their
opinions?

	Trust relationships between consumers, business, military,
intelligence and government are only compatible to a certain point. There
usually is a very good reason for some types of complexity. 

> > 
> > 4. Miscellaneous
> > ================
> > 
> > 4.1 Areas needing further work
> > ==============================
> > - Challenge-response doesn't work with forking (but addresses replay) as
> > only one 401
> > will get back to the caller.
> 
> This can be solved, at the cost of increased proxy complexity, by having
> the proxy gather all challenges from all branches and return them in the
> 401 response, similar to multi-proxy authentication.
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

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<TITLE>RE: [SIP] Minutes of SIP Security task force in Adelaide</TITLE>
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<BODY>
<BR>
<BR>
<UL>
<P><FONT SIZE=3D1 FACE=3D"Arial">-----Original Message-----</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">From:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Henning Schulzrinne =
[SMTP:hgs@cs.columbia.edu]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Saturday, May 13, 2000 3:46 AM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">Jonathan Rosenberg</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Cc:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">sip</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT>=
</B> <FONT SIZE=3D1 FACE=3D"Arial">Re: [SIP] Minutes of SIP Security =
task force in Adelaide</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; A number of people participated =
in the SIP Security Task Force meeting</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Must have been a really secure meeting =
if the identity of the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">participants is secret :-)</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; Jonathan Rosenberg opened the =
discussion by suggesting that:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; - SIP encryption (not =
authentication) should be deprecated, because:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I strongly disagree with this. SIP =
encryption serves a number of useful</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">purposes. It hides information, for =
example, about Contact, Subject,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Organization, Retry-After, media =
types, media IP addresses and meeting</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">times (in SDP), SDP k=3D keys, etc. =
All of these are irrelevant for proxy</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">routing behavior, but highly =
sensitive. </FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">I agree with the =
disagreement.</FONT>=20
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp; o It doesn't work =
with NAT and firewalls</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">That's a risk you take. I refuse to =
dumb down protocols for everybody</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">just because some people are behind =
firewalls and NATs. There are lots</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">of protocols or features that you =
can't use behind a firewall or NAT.</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">One could consider a =
SIP proxy as an ALG and a firewall element, could one not? If you agree =
with this then it does work for NAT and firewalls. </FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">I really sick of =
people using &quot;NAT&quot; as a logical fallacy in their arguments. =
</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Perhaps if we bury =
our heads in the sand long enough they will go away.</FONT>=20
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp; o It assumes you know =
the public key of the party you are calling,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; which is not a good</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; assumption.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">It works in many cases. In other cases =
it can be made to work, e.g., via</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">indirection.</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Yep</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp; o It only really =
hides the body (and again that doesn't work well due</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; to the above).</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">See above.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Initial discussion of these =
points led to the decision that it should</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; still be possible</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; to support end-to-end encryption =
as there may be some uses for it.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Agreed.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; A problem with the current =
end-to-end encryption scheme is, that it</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; relies on knowing the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; public key of the other party. =
One way of solving that is to return a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; public key in the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; response and simply do a =
re-INVITE then. However, it was agreed that we</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; should not do</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; that, as we effectively end up =
duplicating IKE, or something similar.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Dave Oran suggested that =
end-to-end information that needs encryption</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; should simply go in</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; the body (don't encrypt =
headers). We could use multipart for this, or</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; simply allow for a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; single body to be encrypted and =
define an error code to be returned if</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; the proxy needs to</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; see the body (e.g. if SDP is =
encrypted and message traverses a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; firewall). The group</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; decided to accept this =
mechanism. The error code will only support</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; coarse granularity,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; i.e., if multiple bodies are =
encrypted, you will not be told which one</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; is needed in</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; clear, just a general error =
saying "body not understood".</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I had originally (many years ago) =
suggested a similar mechanism,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">assuming I understand your proposal =
correctly, namely to have a body</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">type &quot;SIP message&quot;, which =
is encrypted. A decision was made not to do</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">this (I'm afraid I no longer remember =
the reason; might have been</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">message size concerns or processing =
logic). If you only want to encrypt</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">SDP, I disagree, for the reasons =
stated above. If you want to do both</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">and use the &quot;SIP message&quot; =
body mechanism, I'm not against this, but this</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">does entail a significant protocol =
change, without changing any of the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">other problems.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Additional discussion on =
encryption of headers ensued. Contact is an</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; example of header</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; that can be usefully encrypted =
(end-to-end), but in general, there</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; doesn't seem to be a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; whole lot of use for this =
(generally, if the information is relevant it</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; can't be</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; encrypted). Only want the =
registrar to see the Contact, not any of the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; intermediate</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; proxies.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Again, I disagree, see above.</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">I see a plethora of =
valuable uses for separate encryption process for the Contact and many =
other headers. Especially when there are hierarchies or network =
ownership boundaries between UAs, proxies, registrars, ALGs, =
etc..</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; 3. AUTHENTICATION =
(end-to-end)</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; 3.1 Hop-by-hop =
Authentication</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; We want to have at least one =
mechanism specified in the SIP</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; specification.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Why? This gets into endless arguments =
as to which is better.</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Industry standards =
consortia should pick one for the network they are designing - not the =
protocol. If all of the public infrastructure standards consortia agree =
to use the end to end SA model then that's great. </FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">SIP is a protocol =
that should be flexible enough to support many mechanisms. </FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">For instance I might =
want to use one mechanism for an in-house solution while the public =
infrastructure may warrant something else.</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Then again, a =
military or government organization may wish to use something =
completely different.</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Then again, a =
service provider may way to do something extra if the communications =
are on net.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; It was decided that we should =
have the following hop-by-hop security</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; mechanisms:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; - Server to server: IPSec using =
ESP transport,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; - Client to server: IPSec using =
ESP transport (not clear ESP transport</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; was agreed to)</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Disagree. For practical reasons, TLS =
is much easier to deploy (keying,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">available libraries, available =
infrastructure to obtain keys, etc.). Are</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">there any technical reasons to force =
somebody to implement IPsec?</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">This is but one =
method. Go to a consortia group to get this approved. If a vendor =
doesn't want to support it - then don't.</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">I do agree that this =
is a good and scalable choice for a consortia group building a public =
infrastructure network to make; however, I bet you'll see the other =
stuff used as well if not in addition to. I guess what I'm saying is =
&quot;good start&quot; but this choice is probably a bit too simplistic =
and utopian.</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Many of these =
mechanisms are compatible.</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">How many actual =
providers were at the meeting voicing their opinions?</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Trust relationships =
between consumers, business, military, intelligence and government are =
only compatible to a certain point. There usually is a very good reason =
for some types of complexity.</FONT> </P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; 4. Miscellaneous</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; 4.1 Areas needing further =
work</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; - Challenge-response doesn't =
work with forking (but addresses replay) as</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; only one 401</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; will get back to the =
caller.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">This can be solved, at the cost of =
increased proxy complexity, by having</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">the proxy gather all challenges from =
all branches and return them in the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">401 response, similar to multi-proxy =
authentication.</FONT>
</P>

<P><FONT SIZE=3D2 =
FACE=3D"Arial">_______________________________________________</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">SIP mailing list</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">SIP@lists.bell-labs.com</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://lists.bell-labs.com/mailman/listinfo/sip" =
TARGET=3D"_blank">http://lists.bell-labs.com/mailman/listinfo/sip</A></F=
ONT></U>
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Subject: RE: [SIP] Minutes of SIP Security task force in Adelaide
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The problem with having 25 mechanisms for security is that the probability
of
being able to build a secure system goes down because different vendors make
different choices and thus we get non-interoperability and even the
probability of having any
solution implemented in a given instance goes down.
 
I would prefer to see one less than perfect solution that works most of the
time
then 25 solutions, each perfect for one kind of system.   The existing
mechanisms
aren't close to good enough for such a status.  There are already too many
options.
 
I don't really care what mechanism we choose as as long as we pick one,
and then really work towards getting everyone to implement it.  The present
situation,
which AFAIK there a precious few implementations, and does not provide an
adequate
level of security, is not satisfactory.  Augmenting it with lots more
alternatives is not 
satisfactory.
 
Brian

-----Original Message-----
From: Glenn Morrow [mailto:gmorrow@nortelnetworks.com]
Sent: Monday, May 15, 2000 2:08 PM
To: 'Henning Schulzrinne'; Jonathan Rosenberg
Cc: sip
Subject: RE: [SIP] Minutes of SIP Security task force in Adelaide





	-----Original Message----- 
From:   Henning Schulzrinne [SMTP:hgs@cs.columbia.edu] 
Sent:   Saturday, May 13, 2000 3:46 AM 
To:     Jonathan Rosenberg 
Cc:     sip 
Subject:        Re: [SIP] Minutes of SIP Security task force in Adelaide 

	> A number of people participated in the SIP Security Task Force
meeting 

	Must have been a really secure meeting if the identity of the 
participants is secret :-) 

	> Jonathan Rosenberg opened the discussion by suggesting that: 
> - SIP encryption (not authentication) should be deprecated, because: 

	I strongly disagree with this. SIP encryption serves a number of
useful 
purposes. It hides information, for example, about Contact, Subject, 
Organization, Retry-After, media types, media IP addresses and meeting 
times (in SDP), SDP k= keys, etc. All of these are irrelevant for proxy 
routing behavior, but highly sensitive. 

	I agree with the disagreement. 

	>   o It doesn't work with NAT and firewalls 

	That's a risk you take. I refuse to dumb down protocols for
everybody 
just because some people are behind firewalls and NATs. There are lots 
of protocols or features that you can't use behind a firewall or NAT. 

	One could consider a SIP proxy as an ALG and a firewall element,
could one not? If you agree with this then it does work for NAT and
firewalls. 

	I really sick of people using "NAT" as a logical fallacy in their
arguments. 

	Perhaps if we bury our heads in the sand long enough they will go
away. 

	>   o It assumes you know the public key of the party you are
calling, 
> which is not a good 
> assumption. 

	It works in many cases. In other cases it can be made to work, e.g.,
via 
indirection. 

	Yep 

	>   o It only really hides the body (and again that doesn't work
well due 
> to the above). 

	See above. 

	> 
> Initial discussion of these points led to the decision that it should 
> still be possible 
> to support end-to-end encryption as there may be some uses for it. 

	Agreed. 

	> 
> A problem with the current end-to-end encryption scheme is, that it 
> relies on knowing the 
> public key of the other party. One way of solving that is to return a 
> public key in the 
> response and simply do a re-INVITE then. However, it was agreed that we 
> should not do 
> that, as we effectively end up duplicating IKE, or something similar. 
> 
> Dave Oran suggested that end-to-end information that needs encryption 
> should simply go in 
> the body (don't encrypt headers). We could use multipart for this, or 
> simply allow for a 
> single body to be encrypted and define an error code to be returned if 
> the proxy needs to 
> see the body (e.g. if SDP is encrypted and message traverses a 
> firewall). The group 
> decided to accept this mechanism. The error code will only support 
> coarse granularity, 
> i.e., if multiple bodies are encrypted, you will not be told which one 
> is needed in 
> clear, just a general error saying "body not understood". 

	I had originally (many years ago) suggested a similar mechanism, 
assuming I understand your proposal correctly, namely to have a body 
type "SIP message", which is encrypted. A decision was made not to do 
this (I'm afraid I no longer remember the reason; might have been 
message size concerns or processing logic). If you only want to encrypt 
SDP, I disagree, for the reasons stated above. If you want to do both 
and use the "SIP message" body mechanism, I'm not against this, but this 
does entail a significant protocol change, without changing any of the 
other problems. 

	> 
> Additional discussion on encryption of headers ensued. Contact is an 
> example of header 
> that can be usefully encrypted (end-to-end), but in general, there 
> doesn't seem to be a 
> whole lot of use for this (generally, if the information is relevant it 
> can't be 
> encrypted). Only want the registrar to see the Contact, not any of the 
> intermediate 
> proxies. 

	Again, I disagree, see above. 

	I see a plethora of valuable uses for separate encryption process
for the Contact and many other headers. Especially when there are
hierarchies or network ownership boundaries between UAs, proxies,
registrars, ALGs, etc..

	> 3. AUTHENTICATION (end-to-end) 
> ============================== 

	> 3.1 Hop-by-hop Authentication 
> ============================= 

	> We want to have at least one mechanism specified in the SIP 
> specification. 

	Why? This gets into endless arguments as to which is better. 

	Industry standards consortia should pick one for the network they
are designing - not the protocol. If all of the public infrastructure
standards consortia agree to use the end to end SA model then that's great. 

	SIP is a protocol that should be flexible enough to support many
mechanisms. 

	For instance I might want to use one mechanism for an in-house
solution while the public infrastructure may warrant something else.

	Then again, a military or government organization may wish to use
something completely different. 

	Then again, a service provider may way to do something extra if the
communications are on net. 

	> 
> It was decided that we should have the following hop-by-hop security 
> mechanisms: 
> - Server to server: IPSec using ESP transport, 
> - Client to server: IPSec using ESP transport (not clear ESP transport 
> was agreed to) 

	Disagree. For practical reasons, TLS is much easier to deploy
(keying, 
available libraries, available infrastructure to obtain keys, etc.). Are 
there any technical reasons to force somebody to implement IPsec? 

	This is but one method. Go to a consortia group to get this
approved. If a vendor doesn't want to support it - then don't.

	I do agree that this is a good and scalable choice for a consortia
group building a public infrastructure network to make; however, I bet
you'll see the other stuff used as well if not in addition to. I guess what
I'm saying is "good start" but this choice is probably a bit too simplistic
and utopian.

	Many of these mechanisms are compatible. 

	How many actual providers were at the meeting voicing their
opinions? 

	Trust relationships between consumers, business, military,
intelligence and government are only compatible to a certain point. There
usually is a very good reason for some types of complexity. 

	> 
> 4. Miscellaneous 
> ================ 
> 
> 4.1 Areas needing further work 
> ============================== 
> - Challenge-response doesn't work with forking (but addresses replay) as 
> only one 401 
> will get back to the caller. 

	This can be solved, at the cost of increased proxy complexity, by
having 
the proxy gather all challenges from all branches and return them in the 
401 response, similar to multi-proxy authentication. 

	_______________________________________________ 
SIP mailing list 
SIP@lists.bell-labs.com 
http://lists.bell-labs.com/mailman/listinfo/sip
<http://lists.bell-labs.com/mailman/listinfo/sip>  


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<TITLE>RE: [SIP] Minutes of SIP Security task force in Adelaide</TITLE>

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<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=210043518-15052000>The 
problem with having 25 mechanisms for security is that the probability 
of</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=210043518-15052000>being 
able to build a secure system goes down because different vendors make different 
choices and thus we get non-interoperability and even the probability of having 
any</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=210043518-15052000>solution implemented in a given instance goes 
down.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=210043518-15052000></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=210043518-15052000>I 
would prefer to see one less than perfect solution that works most of the 
time</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=210043518-15052000>then 
25 solutions, each perfect for one kind of system.&nbsp;&nbsp; The existing 
mechanisms</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=210043518-15052000>aren't 
close to good enough for such a status.&nbsp; There are already too many 
options.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=210043518-15052000></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=210043518-15052000>I 
don't really care what mechanism we choose as as long as we pick 
one,</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=210043518-15052000>and 
then really work towards getting everyone to implement it.&nbsp; The present 
situation,</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=210043518-15052000>which 
AFAIK there a precious few implementations, and does not provide an 
adequate</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=210043518-15052000>level 
of security, is not satisfactory.&nbsp; Augmenting </SPAN></FONT><FONT 
color=#0000ff face=Arial size=2><SPAN class=210043518-15052000>it with lots more 
alternatives is not </SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=210043518-15052000>satisfactory.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=210043518-15052000></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=210043518-15052000>Brian</SPAN></FONT></DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Glenn Morrow 
  [mailto:gmorrow@nortelnetworks.com]<BR><B>Sent:</B> Monday, May 15, 2000 2:08 
  PM<BR><B>To:</B> 'Henning Schulzrinne'; Jonathan Rosenberg<BR><B>Cc:</B> 
  sip<BR><B>Subject:</B> RE: [SIP] Minutes of SIP Security task force in 
  Adelaide<BR><BR></DIV></FONT><BR><BR>
  <UL>
    <P><FONT face=Arial size=1>-----Original Message-----</FONT> <BR><B><FONT 
    face=Arial size=1>From:&nbsp;&nbsp;</FONT></B> <FONT face=Arial 
    size=1>Henning Schulzrinne [SMTP:hgs@cs.columbia.edu]</FONT> <BR><B><FONT 
    face=Arial size=1>Sent:&nbsp;&nbsp;</FONT></B> <FONT face=Arial 
    size=1>Saturday, May 13, 2000 3:46 AM</FONT> <BR><B><FONT face=Arial 
    size=1>To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT face=Arial 
    size=1>Jonathan Rosenberg</FONT> <BR><B><FONT face=Arial 
    size=1>Cc:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT face=Arial 
    size=1>sip</FONT> <BR><B><FONT face=Arial 
    size=1>Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT 
    face=Arial size=1>Re: [SIP] Minutes of SIP Security task force in 
    Adelaide</FONT> </P>
    <P><FONT face=Arial size=2>&gt; A number of people participated in the SIP 
    Security Task Force meeting</FONT> </P>
    <P><FONT face=Arial size=2>Must have been a really secure meeting if the 
    identity of the</FONT> <BR><FONT face=Arial size=2>participants is secret 
    :-)</FONT> </P>
    <P><FONT face=Arial size=2>&gt; Jonathan Rosenberg opened the discussion by 
    suggesting that:</FONT> <BR><FONT face=Arial size=2>&gt; - SIP encryption 
    (not authentication) should be deprecated, because:</FONT> </P>
    <P><FONT face=Arial size=2>I strongly disagree with this. SIP encryption 
    serves a number of useful</FONT> <BR><FONT face=Arial size=2>purposes. It 
    hides information, for example, about Contact, Subject,</FONT> <BR><FONT 
    face=Arial size=2>Organization, Retry-After, media types, media IP addresses 
    and meeting</FONT> <BR><FONT face=Arial size=2>times (in SDP), SDP k= keys, 
    etc. All of these are irrelevant for proxy</FONT> <BR><FONT face=Arial 
    size=2>routing behavior, but highly sensitive. </FONT></P>
    <P><FONT color=#0000ff face=Arial size=2>I agree with the 
    disagreement.</FONT> </P>
    <P><FONT face=Arial size=2>&gt;&nbsp;&nbsp; o It doesn't work with NAT and 
    firewalls</FONT> </P>
    <P><FONT face=Arial size=2>That's a risk you take. I refuse to dumb down 
    protocols for everybody</FONT> <BR><FONT face=Arial size=2>just because some 
    people are behind firewalls and NATs. There are lots</FONT> <BR><FONT 
    face=Arial size=2>of protocols or features that you can't use behind a 
    firewall or NAT.</FONT> </P>
    <P><FONT color=#0000ff face=Arial size=2>One could consider a SIP proxy as 
    an ALG and a firewall element, could one not? If you agree with this then it 
    does work for NAT and firewalls. </FONT></P>
    <P><FONT color=#0000ff face=Arial size=2>I really sick of people using "NAT" 
    as a logical fallacy in their arguments. </FONT></P>
    <P><FONT color=#0000ff face=Arial size=2>Perhaps if we bury our heads in the 
    sand long enough they will go away.</FONT> </P>
    <P><FONT face=Arial size=2>&gt;&nbsp;&nbsp; o It assumes you know the public 
    key of the party you are calling,</FONT> <BR><FONT face=Arial size=2>&gt; 
    which is not a good</FONT> <BR><FONT face=Arial size=2>&gt; 
    assumption.</FONT> </P>
    <P><FONT face=Arial size=2>It works in many cases. In other cases it can be 
    made to work, e.g., via</FONT> <BR><FONT face=Arial 
    size=2>indirection.</FONT> </P>
    <P><FONT color=#0000ff face=Arial size=2>Yep</FONT> </P>
    <P><FONT face=Arial size=2>&gt;&nbsp;&nbsp; o It only really hides the body 
    (and again that doesn't work well due</FONT> <BR><FONT face=Arial 
    size=2>&gt; to the above).</FONT> </P>
    <P><FONT face=Arial size=2>See above.</FONT> </P>
    <P><FONT face=Arial size=2>&gt; </FONT><BR><FONT face=Arial size=2>&gt; 
    Initial discussion of these points led to the decision that it should</FONT> 
    <BR><FONT face=Arial size=2>&gt; still be possible</FONT> <BR><FONT 
    face=Arial size=2>&gt; to support end-to-end encryption as there may be some 
    uses for it.</FONT> </P>
    <P><FONT face=Arial size=2>Agreed.</FONT> </P>
    <P><FONT face=Arial size=2>&gt; </FONT><BR><FONT face=Arial size=2>&gt; A 
    problem with the current end-to-end encryption scheme is, that it</FONT> 
    <BR><FONT face=Arial size=2>&gt; relies on knowing the</FONT> <BR><FONT 
    face=Arial size=2>&gt; public key of the other party. One way of solving 
    that is to return a</FONT> <BR><FONT face=Arial size=2>&gt; public key in 
    the</FONT> <BR><FONT face=Arial size=2>&gt; response and simply do a 
    re-INVITE then. However, it was agreed that we</FONT> <BR><FONT face=Arial 
    size=2>&gt; should not do</FONT> <BR><FONT face=Arial size=2>&gt; that, as 
    we effectively end up duplicating IKE, or something similar.</FONT> 
    <BR><FONT face=Arial size=2>&gt; </FONT><BR><FONT face=Arial size=2>&gt; 
    Dave Oran suggested that end-to-end information that needs encryption</FONT> 
    <BR><FONT face=Arial size=2>&gt; should simply go in</FONT> <BR><FONT 
    face=Arial size=2>&gt; the body (don't encrypt headers). We could use 
    multipart for this, or</FONT> <BR><FONT face=Arial size=2>&gt; simply allow 
    for a</FONT> <BR><FONT face=Arial size=2>&gt; single body to be encrypted 
    and define an error code to be returned if</FONT> <BR><FONT face=Arial 
    size=2>&gt; the proxy needs to</FONT> <BR><FONT face=Arial size=2>&gt; see 
    the body (e.g. if SDP is encrypted and message traverses a</FONT> <BR><FONT 
    face=Arial size=2>&gt; firewall). The group</FONT> <BR><FONT face=Arial 
    size=2>&gt; decided to accept this mechanism. The error code will only 
    support</FONT> <BR><FONT face=Arial size=2>&gt; coarse granularity,</FONT> 
    <BR><FONT face=Arial size=2>&gt; i.e., if multiple bodies are encrypted, you 
    will not be told which one</FONT> <BR><FONT face=Arial size=2>&gt; is needed 
    in</FONT> <BR><FONT face=Arial size=2>&gt; clear, just a general error 
    saying "body not understood".</FONT> </P>
    <P><FONT face=Arial size=2>I had originally (many years ago) suggested a 
    similar mechanism,</FONT> <BR><FONT face=Arial size=2>assuming I understand 
    your proposal correctly, namely to have a body</FONT> <BR><FONT face=Arial 
    size=2>type "SIP message", which is encrypted. A decision was made not to 
    do</FONT> <BR><FONT face=Arial size=2>this (I'm afraid I no longer remember 
    the reason; might have been</FONT> <BR><FONT face=Arial size=2>message size 
    concerns or processing logic). If you only want to encrypt</FONT> <BR><FONT 
    face=Arial size=2>SDP, I disagree, for the reasons stated above. If you want 
    to do both</FONT> <BR><FONT face=Arial size=2>and use the "SIP message" body 
    mechanism, I'm not against this, but this</FONT> <BR><FONT face=Arial 
    size=2>does entail a significant protocol change, without changing any of 
    the</FONT> <BR><FONT face=Arial size=2>other problems.</FONT> </P>
    <P><FONT face=Arial size=2>&gt; </FONT><BR><FONT face=Arial size=2>&gt; 
    Additional discussion on encryption of headers ensued. Contact is an</FONT> 
    <BR><FONT face=Arial size=2>&gt; example of header</FONT> <BR><FONT 
    face=Arial size=2>&gt; that can be usefully encrypted (end-to-end), but in 
    general, there</FONT> <BR><FONT face=Arial size=2>&gt; doesn't seem to be 
    a</FONT> <BR><FONT face=Arial size=2>&gt; whole lot of use for this 
    (generally, if the information is relevant it</FONT> <BR><FONT face=Arial 
    size=2>&gt; can't be</FONT> <BR><FONT face=Arial size=2>&gt; encrypted). 
    Only want the registrar to see the Contact, not any of the</FONT> <BR><FONT 
    face=Arial size=2>&gt; intermediate</FONT> <BR><FONT face=Arial size=2>&gt; 
    proxies.</FONT> </P>
    <P><FONT face=Arial size=2>Again, I disagree, see above.</FONT> </P>
    <P><FONT color=#0000ff face=Arial size=2>I see a plethora of valuable uses 
    for separate encryption process for the Contact and many other headers. 
    Especially when there are hierarchies or network ownership boundaries 
    between UAs, proxies, registrars, ALGs, etc..</FONT></P>
    <P><FONT face=Arial size=2>&gt; 3. AUTHENTICATION (end-to-end)</FONT> 
    <BR><FONT face=Arial size=2>&gt; ==============================</FONT> </P>
    <P><FONT face=Arial size=2>&gt; 3.1 Hop-by-hop Authentication</FONT> 
    <BR><FONT face=Arial size=2>&gt; =============================</FONT> </P>
    <P><FONT face=Arial size=2>&gt; We want to have at least one mechanism 
    specified in the SIP</FONT> <BR><FONT face=Arial size=2>&gt; 
    specification.</FONT> </P>
    <P><FONT face=Arial size=2>Why? This gets into endless arguments as to which 
    is better.</FONT> </P>
    <P><FONT color=#0000ff face=Arial size=2>Industry standards consortia should 
    pick one for the network they are designing - not the protocol. If all of 
    the public infrastructure standards consortia agree to use the end to end SA 
    model then that's great. </FONT></P>
    <P><FONT color=#0000ff face=Arial size=2>SIP is a protocol that should be 
    flexible enough to support many mechanisms. </FONT></P>
    <P><FONT color=#0000ff face=Arial size=2>For instance I might want to use 
    one mechanism for an in-house solution while the public infrastructure may 
    warrant something else.</FONT></P>
    <P><FONT color=#0000ff face=Arial size=2>Then again, a military or 
    government organization may wish to use something completely 
    different.</FONT> </P>
    <P><FONT color=#0000ff face=Arial size=2>Then again, a service provider may 
    way to do something extra if the communications are on net.</FONT> </P>
    <P><FONT face=Arial size=2>&gt; </FONT><BR><FONT face=Arial size=2>&gt; It 
    was decided that we should have the following hop-by-hop security</FONT> 
    <BR><FONT face=Arial size=2>&gt; mechanisms:</FONT> <BR><FONT face=Arial 
    size=2>&gt; - Server to server: IPSec using ESP transport,</FONT> <BR><FONT 
    face=Arial size=2>&gt; - Client to server: IPSec using ESP transport (not 
    clear ESP transport</FONT> <BR><FONT face=Arial size=2>&gt; was agreed 
    to)</FONT> </P>
    <P><FONT face=Arial size=2>Disagree. For practical reasons, TLS is much 
    easier to deploy (keying,</FONT> <BR><FONT face=Arial size=2>available 
    libraries, available infrastructure to obtain keys, etc.). Are</FONT> 
    <BR><FONT face=Arial size=2>there any technical reasons to force somebody to 
    implement IPsec?</FONT> </P>
    <P><FONT color=#0000ff face=Arial size=2>This is but one method. Go to a 
    consortia group to get this approved. If a vendor doesn't want to support it 
    - then don't.</FONT></P>
    <P><FONT color=#0000ff face=Arial size=2>I do agree that this is a good and 
    scalable choice for a consortia group building a public infrastructure 
    network to make; however, I bet you'll see the other stuff used as well if 
    not in addition to. I guess what I'm saying is "good start" but this choice 
    is probably a bit too simplistic and utopian.</FONT></P>
    <P><FONT color=#0000ff face=Arial size=2>Many of these mechanisms are 
    compatible.</FONT> </P>
    <P><FONT color=#0000ff face=Arial size=2>How many actual providers were at 
    the meeting voicing their opinions?</FONT> </P>
    <P><FONT color=#0000ff face=Arial size=2>Trust relationships between 
    consumers, business, military, intelligence and government are only 
    compatible to a certain point. There usually is a very good reason for some 
    types of complexity.</FONT> </P>
    <P><FONT face=Arial size=2>&gt; </FONT><BR><FONT face=Arial size=2>&gt; 4. 
    Miscellaneous</FONT> <BR><FONT face=Arial size=2>&gt; 
    ================</FONT> <BR><FONT face=Arial size=2>&gt; </FONT><BR><FONT 
    face=Arial size=2>&gt; 4.1 Areas needing further work</FONT> <BR><FONT 
    face=Arial size=2>&gt; ==============================</FONT> <BR><FONT 
    face=Arial size=2>&gt; - Challenge-response doesn't work with forking (but 
    addresses replay) as</FONT> <BR><FONT face=Arial size=2>&gt; only one 
    401</FONT> <BR><FONT face=Arial size=2>&gt; will get back to the 
    caller.</FONT> </P>
    <P><FONT face=Arial size=2>This can be solved, at the cost of increased 
    proxy complexity, by having</FONT> <BR><FONT face=Arial size=2>the proxy 
    gather all challenges from all branches and return them in the</FONT> 
    <BR><FONT face=Arial size=2>401 response, similar to multi-proxy 
    authentication.</FONT> </P>
    <P><FONT face=Arial 
    size=2>_______________________________________________</FONT> <BR><FONT 
    face=Arial size=2>SIP mailing list</FONT> <BR><FONT face=Arial 
    size=2>SIP@lists.bell-labs.com</FONT> <BR><U><FONT color=#0000ff face=Arial 
    size=2><A href="http://lists.bell-labs.com/mailman/listinfo/sip" 
    target=_blank>http://lists.bell-labs.com/mailman/listinfo/sip</A></FONT></U> 
    </P></UL></BLOCKQUOTE></BODY></HTML>

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To: Giuseppe Ricagni <giuseppe.ricagni@italtel.it>
Cc: Michael Thomas <mat@cisco.com>, James Kempf <James.Kempf@Eng.Sun.COM>,
        sip@lists.bell-labs.com, jdrosen@dynamicsoft.com
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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Giuseppe Ricagni writes:
 >    * possible optimizations (e.g.: signalling for a basic call to an Australian
 >      fixed network number from an Italian mobile roaming to Australia could be
 >      processed locally, without any "tromboning")

   If during call setup there needs to be a round
   trip to the home network, it pretty much negates
   any advantage of proxying the call locally. 

 >    * some people claim the "proxy" should also perform some kind of admission
 >      control for the calls. The proxy in the home domain has no information on the
 >      visited domain

   This is true of the foreign domain too. How does
   any proxy have knowledge of network topology? 

 >    * the PNO who runs the visited network might desire to have some kind of
 >      knowledge on what is going on in his network, from a call control point of
 >      view

   Absolutely. However, admission control can
   be done through other means. Namely, you could
   use RSVP to signal your intent to consume some
   air resources and authenticate/authorize that
   traffic. SIP normally has no part in bandwidth
   allocation, which is a feature not a bug. 
   Even if you believe the manyfolks (DQoS) model
   of implicit linkage, I still don't see why
   there is any advantage to having a local proxy
   do call routing vs my home proxy. I can think
   of a lot of disadvantages though.

	Mike


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Subject: [SIP] 2nd CFP: IP Telecom Services Workshop 2000
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[Please accept our apologies for multiple copies of this call.]

-------------  SECOND CALL FOR PAPERS  ---------------

       IP TELECOM SERVICES WORKSHOP 2000 (IPTS 2000)

                    co-located with
             Fall 2000 Voice on the Net (VON)

                     organized by
                  AT&T Labs Research
                      pulver.com

         Cobb Galleria, Atlanta, Georgia, U.S.A
                  September 11, 2000

            submission deadline: June 16, 2000
         http://www.research.att.com/conf/ipts2000/

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

IPTS 2000 is a new workshop dedicated to the important, emerging field of IP
Telecom Services research.

In contrast to the services available on the PSTN, a new world of telecom
services is possible on an IP-based infrastructure. This world presents new
challenges for service development, deployment and management. This one-day
workshop will serve as a forum for the dissemination of research relating to
these challenges.

The IPTS 2000 Program Committee invites submission of papers on substantial,
original and previously unpublished research in IP telecom services, including,
but not limited to:

- service architectures
  (e.g. AIN, SoftSwitch, JAIN, Parlay, DFC)
- service creation environments
- protocols, standards and interoperability
  (e.g. H.323, SIP, MGCP, Megaco/H.248, etc.)
- service programming languages
- service management - billing, fault tolerance, monitoring
- novel services

We welcome papers describing original research; position papers describing
research interests in the field, work in progress, or future directions of
research; project or system descriptions.


IMPORTANT DATES

June 16, 2000        Submission deadline
July 21, 2000        Notification of acceptance
August 25, 2000      Final version of papers
September 11, 2000   IPTS 2000


PAPER SUBMISSIONS

Papers must describe original, previously unpublished work that has not been
simultaneously submitted for publication elsewhere. They must be written in
English, not exceeding 6 pages including figures, tables and references in 10-12
point font on 8.5x11-inch (US letter) paper. The first page must include an
abstract of up to 200 words, keywords, and email address of the corresponding
author.

All papers must be submitted electronically unless specifically approved by the
Workshop Co-Chairs. Submissions in PostScript or PDF format should be sent to
mailto:ipts2000@research.att.com.

Authors of accepted papers are expected to present their contribution at the
workshop.  The proceedings will be published in hardcopy and on the web.


REVIEW PROCEDURE

All submissions will be subject to academic peer review by the IPTS 2000 Program
Committee under the chairmanship of the Workshop Co-Chairs. Each paper will
receive three written reviews, and reviews will be returned to the authors.  The
Workshop Co-Chairs have final authority over the review process and all
decisions relating to acceptance of papers. Review criteria include originality
of ideas, technical soundness, significance of results, and quality of
presentation.


CONFERENCE VENUE AND RELATED EVENTS

IPTS 2000 will be held on September 11, 2000 in Atlanta, Georgia. The workshop
will be co-located with Fall 2000 Voice on the Net (VON).  Details about Fall
2000 VON can be found at http://pulver.com.

IPTS 2000 is organized by AT&T Labs Research (http://www.research.att.com) and
pulver.com (http://pulver.com).


WORKSHOP CO-CHAIRS

Greg Bond, AT&T Labs Research
Eric Cheung, AT&T Labs Research

PROGRAM COMMITTEE

Mauricio Arango, Sun Microsystems Labs
Joanne Atlee, University of Waterloo
Ralph Blumenthal, Daewoo Telecom
Scott Hoffpauir, BroadSoft
Evan Magill, University of Strathclyde
Peter Mataga, dynamicsoft
Bernie Pagurek, Carleton University
Jonathan Rosenberg, dynamicsoft
Henning Schulzrinne, Columbia University
Greg Utas, Nortel Networks
Pamela Zave, AT&T Labs Research


FURTHER GENERAL INFORMATION

http://www.research.att.com/conf/ipts2000


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From sip-admin@lists.bell-labs.com  Mon May 15 17:08:12 2000
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To: Glenn Morrow <gmorrow@nortelnetworks.com>
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Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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> Glenn Morrow wrote:
> 
>      > Jonathan Rosenberg opened the discussion by suggesting that:
>      > - SIP encryption (not authentication) should be deprecated,
>      because:
> 
>      I strongly disagree with this. SIP encryption serves a number of
>      useful
>      purposes. It hides information, for example, about Contact,
>      Subject,
>      Organization, Retry-After, media types, media IP addresses and
>      meeting
>      times (in SDP), SDP k= keys, etc. All of these are irrelevant for
>      proxy
>      routing behavior, but highly sensitive.
> 
>      I agree with the disagreement.

There isn't really any disagreement. I made this obviously bold
statement as an expression of my concerns with relying on end to end as
the sole mechanism for privacy. In the end, we did agree that it should
stay.
 

> 
>      >   o It doesn't work with NAT and firewalls
> 
>      That's a risk you take. I refuse to dumb down protocols for
>      everybody
>      just because some people are behind firewalls and NATs. There are
>      lots
>      of protocols or features that you can't use behind a firewall or
>      NAT.
> 
>      One could consider a SIP proxy as an ALG and a firewall element,
>      could one not? If you agree with this then it does work for NAT
>      and firewalls.

No, this does *not* work. The body of the SIP message is encrypted with
the public key of the target in the To field. This is therefore not
readable by anyone except the recipient, which means that NAT,
co-located with a proxy or otherwise, can't see it. Normal proxies don't
need to see it, but an ALG does. Thats why SIP doesn't work through NATs
when encrypted. But, hop by hop does work, since the proxy can decrypt
the message.

Anyway, I believe there are uses for e2e encryption. Like everything
else in engineering, its a tradeoff. Go for top notch security, and you
get no services. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Henning Schulzrinne wrote:
> 
> > 4.1 Areas needing further work
> > ==============================
> > - Challenge-response doesn’t work with forking (but addresses replay) as
> > only one 401
> > will get back to the caller.
> 
> This can be solved, at the cost of increased proxy complexity, by having
> the proxy gather all challenges from all branches and return them in the
> 401 response, similar to multi-proxy authentication.

This is an interesting idea. I like it. An interesting question arises
when you get *both* 401s and 407s. I guess the proxy could return a 401,
and place into the 401 WWW-Authenticate headers from the 401s it
received, and Proxy-Authenticate headers from the 407s. A UA would need
to be smart and look for both headers in a 401, however. Any other
interactions...?

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Mon May 15 18:23:59 2000
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Rather than do nesting of >, maybe we can make progress by summarizing,
as several different problems seem to be discussed at the same time:

- e2e authentication and/or encryption across domains:

Useful primarily for authenticated call filtering and key exchange for
media encryption. Scalable probably for intelligent end systems, at the
possible penalty of call setup delay. If truly e2e, requires per-user
public keys, which do not seem imminent (although some organizations are
apparently working on this, mostly for internal purchasing use).
Semi-e2e might use Kerberos, local proxy digest authentication or other
h2h mechanism (see below) to authenticate and secure the local hop, with
proxy-secured signaling. This only requires a PK for each organization,
readily obtained from Verisign and their various competitors that would
be recognized in web browsers. This would essentially be a model similar
to today's web model, which seems to work satisfactorily in the absence
of a true PKI. Proxy-based security probably scales easily (in the
typical CPU-dedicated-to-server installation) to a few thousand calls a
day (i.e., it would work for typical departments), based on rough
assumptions of SSL servers. (Is there an SSL/TLS benchmark test for web
servers?) [TLS plays no role here, just probably has similar
performance.]

- h2h mechanisms *across* domains

This seems to be the hard problem. Just specifying IPsec without an IKE
doesn't help much to achieve interoperability. Is it even plausible to
run basically every single IKE version and variation on a proxy, to be
able to talk to every other domain? Would PKCROSS be sufficiently
acceptable to every other proxy operator to use it exclusively? Is there
operational experience with large-scale use of this? The number of
public key operations would seem to be the same with PKCROSS or direct
proxy-to-proxy use of some IKE or TLS, depending solely on the sparsity
of proxy relationships. PKCROSS would presumably have the advantage of
splitting this across two (sets of) machines, but unless this is used
for something else much "bigger" and can thus ride the coattails of an
existing application, the net number of boxes seems comparable.

For TLS, while restricted to TCP, there's operational experience and a
well-known infrastructure for obtaining certificates; most organizations
would already have such a certificate for their secure web server.

This problem seems much harder than that faced by typical sigtran or
MGCP/Megaco applications, where (for Megaco) this is mostly intradomain
or for sigtran a smallish number of well-known peers instead of every
ISP and organization-with-a-SIP-address on the planet.

Thus, the scalability depends on the fraction of SIP calls that require
SA setup. One could presumably look at mail server logs to see if, say,
80% of the messages go to 100 domains with lots of traffic per PK
operation, so that only 20% would require per-call SA establishment.
Besides the usual 80-20 rule, I have no indications what, if any,
scaling advantage will accrue due to such uneven distribution of call
destinations.

- h2h *within* a domain

Relatively easy, since one can agree on either a single IKE or (most
likely currently) IKE implementation, with alternatives such as Kerberos
or just about anything else feasible according to what's available
locally. (E.g., Microsoft-only shops might use the Microsoft flavor and
not care about any interoperability problems with the rest of the world.
Or, for authentication only, somebody might draw on CHAP
authentication.) Specifying a single or small number of architectures is
probably helpful, but to a somewhat more limited extent than in the
inter-domain case, since single organizations usually have a mechanism
to agree on things. Specifying Kerberos seems like an easy step forward
here. If applications use GSSAPI, this is probably largely invisible to
the application writer.

Henning




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Date: Mon, 15 May 2000 18:30:23 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: "Rosen, Brian" <brosen@fore.com>
Cc: "'Glenn Morrow'" <gmorrow@nortelnetworks.com>,
        "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        sip <sip@lists.bell-labs.com>
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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> The problem with having 25 mechanisms for security is that the probability of
> being able to build a secure system goes down because different vendors make different choices and thus we get
> non-interoperability and even the probability of having any
> solution implemented in a given instance goes down.
>  
> I would prefer to see one less than perfect solution that works most of the time
> then 25 solutions, each perfect for one kind of system.   The existing mechanisms
> aren't close to good enough for such a status.  There are already too many options.
>  
> I don't really care what mechanism we choose as as long as we pick one,
> and then really work towards getting everyone to implement it.  The present situation,
> which AFAIK there a precious few implementations, and does not provide an adequate
> level of security, is not satisfactory.  Augmenting it with lots more alternatives is not 
> satisfactory.

I have no problems with alternatives if there is some kind of baseline
that everyone does. The situation is similar with codecs; there are lots
of choices, and sometimes people specify a baseline. Interestingly,
SIP/SDP has never specified a baseline codec, nor can we, I don't think.
Every seems to do G.711, so things work. We all know the fun involved in
standardizing baseline low rate codecs...

So, the question to me is this: should we, or need we, specify baseline
(i.e., mandatory if you do security at all) mechanisms for e2e and hop
by hop, or do we let the industry effectively pick a defacto? Brian has
argued that letting the industry pick one will lead to interop problems,
since it effectively might not pick one. Others will argue that we will
never be able to pick one, since there is no clear winner.

I don't have an easy answer for this, since its more
business/political/procedural than it is technical. I would actually
look to the ADs for input; is it necessary for an application layer
protocol to pick baseline mandatory-to-implement security mechanisms?


-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Mon May 15 19:01:33 2000
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Let me chime in with strong support of what Mike is saying. One of the
great strengths of SIP is that there is a complete separation of
transport (i.e.,voice) from signaling (read: services). This means that
I can buy my transport and access for VoIp from one provider, but get
applications and services from one or more other providers. This provide
disaggregation of what has traditionally been a coupled service
offering. Such disaggregation opens the door to massive innovation in
the services area, since any yahoo can buy application hosting services,
run on application on the hosted providers SIP servers, and provide
service to anyone in the world, accessing from any IP device, wireless
included.

This is the model that made the web successful. Would it had evolved if
ONLY my ISP could provide web content? No way. There would *never* had
been a web if content and services for web had been coupled to access
and transport. Never. Please lets not make this mistake with
communications services.

Efforts to link transport and access with services and signaling on the
Internet are fundamentally futile, flawed, and will, in the end, reduce
competition and innovation. But, I suppose thats the point.

So, as it relates to wireless, just because some local provider is
offering me transport services does not mean I should be forced to use
them for signaling and features. 

Michael Thomas wrote:
> 
> Giuseppe Ricagni writes:
>  >    * possible optimizations (e.g.: signalling for a basic call to an Australian
>  >      fixed network number from an Italian mobile roaming to Australia could be
>  >      processed locally, without any "tromboning")
> 
>    If during call setup there needs to be a round
>    trip to the home network, it pretty much negates
>    any advantage of proxying the call locally.

Please note that this service (optimized dial plans based on roaming) is
still possible even if I go directly to the home network. All I need to
do is indicate to my home proxy where I am, so it knows what dial plan
to apply. In fact, dial plan administration is, in general, a service
that can be outsourced to any ASP (of course, only local ASPs may wish
to provide it, but the point is anyone can).

> 
>  >    * some people claim the "proxy" should also perform some kind of admission
>  >      control for the calls. The proxy in the home domain has no information on the
>  >      visited domain
> 
>    This is true of the foreign domain too. How does
>    any proxy have knowledge of network topology?

Call admission control in proxies >>will never work<<. As Mike suggests,
this belongs in the transport layer, not the signaling layer.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Tue May 16 03:57:25 2000
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Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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> No, this does *not* work. The body of the SIP message is encrypted with
> the public key of the target in the To field. This is therefore not
> readable by anyone except the recipient, which means that NAT,
> co-located with a proxy or otherwise, can't see it. Normal proxies don't
> need to see it, but an ALG does. Thats why SIP doesn't work through NATs
> when encrypted. But, hop by hop does work, since the proxy can decrypt
> the message.

The NAT issue is more general, in that we could imagine, for example, a
different payload format than SDP, say for event notification or as
SDPng. In all such cases, the ALG would have to return some status code
and refuse the call or hope that there are no IP addresses in the part
hidden (after all, if the inside-NAT-host is send-only, it wouldn't much
matter - not a course of action I'd recommend). It then becomes the
decision of the parties involved whether to complain to the NAT/ALG
builder, get the admin that got the company into NATs fired or reduce
their expectation of the level of service.




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From sip-admin@lists.bell-labs.com  Tue May 16 08:04:23 2000
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>Giuseppe Ricagni writes:
> >    * possible optimizations (e.g.: signalling for a basic call to an
>Australian
> >      fixed network number from an Italian mobile roaming to Australia could
>be
> >      processed locally, without any "tromboning")
>
>   If during call setup there needs to be a round
>   trip to the home network, it pretty much negates
>   any advantage of proxying the call locally. 
>

When you take roaming into account, there is an advantage. Suppose the
Italian now in Australia moves. If the home proxy is used, the network
has to go back to the home to roam. If a local proxy is used, there is
reduced latency. With a local proxy, you pay the round trip cost
to the home network just once.

In addition, some people have been talking about using SIP for other
kinds of in-call signalling. If the mobile has to go all the way
back to the home proxy, there is simply too much latency.






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From sip-admin@lists.bell-labs.com  Tue May 16 08:39:05 2000
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Jonathan,

Unfortunately, I have to respectfully disagree.

I don't think the analogy between SIP and HTTP is valid. I see it more
as being between SIP and routing. Suppose I would have to go all the
way back to my home network to get my packets routed. 
I view SIP as being a way of doing the kind of things that routing protocols
do, except at the application layer. The network of SIP proxies is performing
telephone call routing.

Even if you are willing to grant the analogy, HTTP requests nowadays often
don't get routed to the server that somebody set up as their web site.
If your ISP is concerned about service, they may have set up an Inktomi
cache, and if the publisher is concerned, they may have contracted with
Akami to do the same. So the world is not so simple as "let's make it
free and easy for anybody to set up service" on the Web anymore. You can
set up your service, but unless you have a concerned ISP or the money
to pay Akami, the latency in using your service may cause people to
go elsewhere.

I know this isn't going to be a popular viewpoint, but I think we need
to be realistic about the design. SIP already contains support for
telling an incoming caller that a client has moved, I can't see the
harm in using a remote proxy, given the performance concerns.

		jak 


>Let me chime in with strong support of what Mike is saying. One of the
>great strengths of SIP is that there is a complete separation of
>transport (i.e.,voice) from signaling (read: services). This means that
>I can buy my transport and access for VoIp from one provider, but get
>applications and services from one or more other providers. This provide
>disaggregation of what has traditionally been a coupled service
>offering. Such disaggregation opens the door to massive innovation in
>the services area, since any yahoo can buy application hosting services,
>run on application on the hosted providers SIP servers, and provide
>service to anyone in the world, accessing from any IP device, wireless
>included.
>
>This is the model that made the web successful. Would it had evolved if
>ONLY my ISP could provide web content? No way. There would *never* had
>been a web if content and services for web had been coupled to access
>and transport. Never. Please lets not make this mistake with
>communications services.
>
>Efforts to link transport and access with services and signaling on the
>Internet are fundamentally futile, flawed, and will, in the end, reduce
>competition and innovation. But, I suppose thats the point.
>
>So, as it relates to wireless, just because some local provider is
>offering me transport services does not mean I should be forced to use
>them for signaling and features. 
>
>Michael Thomas wrote:
>> 
>> Giuseppe Ricagni writes:
>>  >    * possible optimizations (e.g.: signalling for a basic call to an
>Australian
>>  >      fixed network number from an Italian mobile roaming to Australia could
>be
>>  >      processed locally, without any "tromboning")
>> 
>>    If during call setup there needs to be a round
>>    trip to the home network, it pretty much negates
>>    any advantage of proxying the call locally.
>
>Please note that this service (optimized dial plans based on roaming) is
>still possible even if I go directly to the home network. All I need to
>do is indicate to my home proxy where I am, so it knows what dial plan
>to apply. In fact, dial plan administration is, in general, a service
>that can be outsourced to any ASP (of course, only local ASPs may wish
>to provide it, but the point is anyone can).
>
>> 
>>  >    * some people claim the "proxy" should also perform some kind of
>admission
>>  >      control for the calls. The proxy in the home domain has no information
>on the
>>  >      visited domain
>> 
>>    This is true of the foreign domain too. How does
>>    any proxy have knowledge of network topology?
>
>Call admission control in proxies >>will never work<<. As Mike suggests,
>this belongs in the transport layer, not the signaling layer.
>
>-Jonathan R.
>-- 
>Jonathan D. Rosenberg                       72 Eagle Rock Ave.
>Chief Scientist                             First Floor
>dynamicsoft                                 East Hanover, NJ 07936
>jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
>http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
>http://www.dynamicsoft.com
>
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip




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As far as whether to talk to the home proxy or local proxy, for a wireless
roaming user which is  changing domains frequently during mid-session-call and
is on the move, it may be better to talk to the local proxy rather than going to
the home proxy all the way during every hand-off (for latency reason I guess),
of course there needs to be a security association between the local-proxy and
home proxy or via public-proxy. This is one of the things we were trying to
debate during SIP-2000 conference in Paris last week to see the trade-off
between either talking to home-proxy or local domain proxy during frequent
hand-off, then there is also a possibility of interaction between local SIP
server and AAA server for a roaming user's profile verification during domain
handoff.

Thanks
Ashutosh




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Date: Tue, 16 May 2000 14:53:14 +0200 (CEST)
From: Lars Berggren <lars@intertex.se>
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Subject: [SIP] Comments on Session Timer draft
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After reading the draft-ietf-sip-session-timer-01.txt I have some
comments.

For the Session Timer to be applied for a call leg, the specifications in
the draft requires that both the UAC and UAS understand the extension
(See example 8.5). However, since the keep-alive mechanism is achieved by
sending 'standard' re-INVITEs, it is sufficient (and useful) that at least
one of the parties is aware of the Session Timer to accomplish a
keep-alive mechanism. The usefulness of such an approach is for example
when a UAC supports 'timer', a stateful proxy server, SPS, wants to apply
a Session Timer to the call and the UAS doesn't understand the extension.
Consider the example 8.5 in the draft:

>   Calling UA -> SPS
>          INVITE
>          k: timer
>
>   SPS -> Called UA
>          INVITE                    SPS adds S-E header
>          k: timer
>          Session-Expires: 180
>
>   SPS <- Called UA
>          200 OK            Called UA doesn't understand session timer
>
>   Calling UA <- SPS
>          200 OK
>
>   Calling UA -> SPS
>          ACK
>
>   SPS -> Called UA
>          ACK

Here, the call will be setup without Session Timer even though the SPS
wanted one and the UAC could do the necessary re-inviting. The UAS doesn't
need to know about Session Timers in order to process re-INVITEs properly.
The only thing that forces that the UAS have to be aware of the Session
Timer is the statement in the draft that the UAS SHOULD send BYE if no
re-INVITE has arrived when the session expires. Well, the UAC could do the
sending of the BYE if it does not intend to update the timer.

So, to solve this problem, I would like to propose that the SPS could add
a Session-Expires header in the response if the SPS knows the UAC supports
'timer' and the response does not contain a Session-Expires header.

Another similar situation occurs when the UAC does not support 'timer' but
the UAS does. Then, according to the draft, the SPS MUST NOT add a
Session-Expires header. If the SPS were allowed to add such a header also
in this case, the UAS could do the refreshes and the 'timer' would be
applied. The UAS would simply look at the Supported header, if present. If
the request states support for timer, the UAS knows the UAC does the
re-inviting, otherwise the UAS will do it.

A useful semantic that the proxy can signal with this proposal is "I'm
giving you a service here, but only for a limited period of time if you do
not request to refresh the service". As opposed to the current draft that
limits the SPS expressivness to "I require that you support session timer
to get my service" or "Please, use the session timer".

Apologies if I have missed something fundamental.

/Lars
--
Lars Berggren       <mailto:lars.berggren@intertex.se>
Intertex Data AB    tel: +46-8-6282828
Sundbyberg, Sweden  fax: +46-8-6286414



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From sip-admin@lists.bell-labs.com  Tue May 16 10:21:25 2000
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From: "Glenn Morrow" <gmorrow@nortelnetworks.com>
To: sip <sip@lists.bell-labs.com>
Subject: RE: [SIP] Minutes of SIP Security task force in Adelaide
Date: Tue, 16 May 2000 09:07:05 -0500
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I don't believe people are opposed to the proposal by the security task
force for the basic public infrastructure. 

I believe the concern is the verbage wanting to get rid of support for the
other methods of security such as a separate encryption of headers and
values at the application level. 

The primary reason I see for this back-pressure being that the trust model
only goes so far for certain types of organizations and individuals. Please
take a look at say a STU-III terminal and it's requirements.

To remove these built-in mechanisms would undermine a rather significant
advantage of SIP over other signaling methods. 

Respectfully,

Glenn

++++++++++++++++++++++++++++

The problem with having 25 mechanisms for security is that the probability
of
being able to build a secure system goes down because different vendors make
different choices and thus we get non-interoperability and even the
probability of having any
solution implemented in a given instance goes down.
 
I would prefer to see one less than perfect solution that works most of the
time
then 25 solutions, each perfect for one kind of system.   The existing
mechanisms
aren't close to good enough for such a status.  There are already too many
options.
 
I don't really care what mechanism we choose as as long as we pick one,
and then really work towards getting everyone to implement it.  The present
situation,
which AFAIK there a precious few implementations, and does not provide an
adequate
level of security, is not satisfactory.  Augmenting it with lots more
alternatives is not 
satisfactory.
 
Brian




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<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2651.65">
<TITLE>RE: [SIP] Minutes of SIP Security task force in Adelaide</TITLE>
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<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">I don't believe =
people are opposed to the proposal by the security task force for the =
basic public infrastructure. </FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">I believe the =
concern is the verbage wanting to get rid of support for the other =
methods of security such as a separate encryption of headers and values =
at the application level. </FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">The primary reason I =
see for this back-pressure being that the trust model only goes so far =
for certain types of organizations and individuals. Please take a look =
at say a STU-III terminal and it's requirements.</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">To remove these =
built-in mechanisms would undermine a rather significant advantage of =
SIP over other signaling methods. </FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Respectfully,</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Glenn</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">++++++++++++++++++++++++++++</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">The problem with =
having 25 mechanisms for security is that the probability of</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">being able to build =
a secure system goes down because different vendors make different =
choices and thus we get non-interoperability and even the probability =
of having any</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">solution implemented =
in a given instance goes down.</FONT>
<BR><FONT FACE=3D"Arial">&nbsp;</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">I would prefer to =
see one less than perfect solution that works most of the time</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">then 25 solutions, =
each perfect for one kind of system.&nbsp;&nbsp; The existing =
mechanisms</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">aren't close to =
good enough for such a status.&nbsp; There are already too many =
options.</FONT>
<BR><FONT FACE=3D"Arial">&nbsp;</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">I don't really care =
what mechanism we choose as as long as we pick one,</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">and then really =
work towards getting everyone to implement it.&nbsp; The present =
situation,</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">which AFAIK there a =
precious few implementations, and does not provide an adequate</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">level of security, =
is not satisfactory.&nbsp; Augmenting it with lots more alternatives is =
not</FONT>=20
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">satisfactory.</FONT>
<BR><FONT FACE=3D"Arial">&nbsp;</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Brian</FONT>
</P>
<BR>
<BR>

</BODY>
</HTML>
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From sip-admin@lists.bell-labs.com  Tue May 16 11:11:57 2000
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Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Michael Thomas" <mat@cisco.com>, <sip@lists.bell-labs.com>,
        "Giuseppe Ricagni" <giuseppe.ricagni@italtel.it>
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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James Kempf writes:
 > I don't think the analogy between SIP and HTTP is valid. I see it more
 > as being between SIP and routing. Suppose I would have to go all the
 > way back to my home network to get my packets routed. 

   This is a pretty dubious analogy. We're talking
   about proxy-routed signaling traffic and that
   alone. You're making it sound like the media
   would do dog-leg routing through the home
   network which is absurd.

 > Even if you are willing to grant the analogy, HTTP requests nowadays often
 > don't get routed to the server that somebody set up as their web site.
 > If your ISP is concerned about service, they may have set up an Inktomi
 > cache, and if the publisher is concerned, they may have contracted with
 > Akami to do the same. So the world is not so simple as "let's make it
 > free and easy for anybody to set up service" on the Web anymore. 

   You're assuming that every thing is going to
   be a gigantic commercial service. I disagree. 
   It's still perfectly possible for me to run a
   web server at home, and in fact I do. We
   haven't engineered walls around that
   possibility. Requiring that I use a foreign
   proxy does, or at least thinks it does.

   Also: you're talking about transparent caching,
   not proxying in the web case. I've not heard
   of any ISP that *require* that I go through 
   their HTTP proxies.

 > You can
 > set up your service, but unless you have a concerned ISP or the money
 > to pay Akami, the latency in using your service may cause people to
 > go elsewhere.

   This analogy is fatally flawed. Cache's are 
   great for the mountains of .gif's. There aren't
   similar considerations for SIP. Also: even
   if you're talking about co-lo servers
   distributed geographically, the site owner
   still "owns" the site. What you're talking
   about here is something run by their
   *competition* on your subscriber's behalf.

   Yuck.
 
 > I know this isn't going to be a popular viewpoint, but I think we need
 > to be realistic about the design. SIP already contains support for
 > telling an incoming caller that a client has moved, I can't see the
 > harm in using a remote proxy, given the performance concerns.

   I think we have to demonstrate that there
   are legitimate performance concerns before we
   solve for them.

   Sanity check: we're talking about a
   hypothetical case where the SIP UA is
   in the handset, right? We're not talking
   about the case where you're terminating SIP
   at the next logical hop up (BTS, MSC) for
   legacy, right? I'm talking about the former.
   In the latter, its debatable which proxy
   should do routing.

	   Mike


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From sip-admin@lists.bell-labs.com  Tue May 16 11:31:08 2000
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To: "Ashutosh Dutta" <adutta@telcordia.com>
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Ashutosh Dutta writes:
 > As far as whether to talk to the home proxy or local proxy, for a wireless
 > roaming user which is  changing domains frequently during mid-session-call and
 > is on the move, it may be better to talk to the local proxy rather than going to
 > the home proxy all the way during every hand-off (for latency reason I guess),
 > of course there needs to be a security association between the local-proxy and
 > home proxy or via public-proxy. This is one of the things we were trying to
 > debate during SIP-2000 conference in Paris last week to see the trade-off
 > between either talking to home-proxy or local domain proxy during frequent
 > hand-off, then there is also a possibility of interaction between local SIP
 > server and AAA server for a roaming user's profile verification during domain
 > handoff.

   Again, sanity check: are we talking about 
   a SIP UA in the handset? If so, I don't see
   why an established call needs to go through
   *any* proxy. Handing off is a matter of
   reestablishing base level (IP) connectivity
   which means AAA to the local provider and
   probably some RSVP futzing to do the media
   handoff.

   If we're talking about SIP'ed legacy handoff,
   we're talking about UA to UA handoff, not proxy
   to proxy. As I said in previous mail, it's less
   clear to me which is better in that case. 
   Feature transparency may be better if you route
   it through your home proxy, but you still have
   a fundamental problem with discordant
   UA/Proxies and the general database problem.

   This is *not* a problem for the UA in the
   handset though; introducing a foreign proxy
   takes us right back to that feature
   transparency problem, which to me seems like
   a huge mistake.

	  Mike


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	Below 


> > Glenn Morrow wrote:
> > 
> >      > Jonathan Rosenberg opened the discussion by suggesting that:
> >      > - SIP encryption (not authentication) should be deprecated,
> >      because:
> > 
> >      I strongly disagree with this. SIP encryption serves a number of
> >      useful
> >      purposes. It hides information, for example, about Contact,
> >      Subject,
> >      Organization, Retry-After, media types, media IP addresses and
> >      meeting
> >      times (in SDP), SDP k= keys, etc. All of these are irrelevant for
> >      proxy
> >      routing behavior, but highly sensitive.
> > 
> >      I agree with the disagreement.
> 
> There isn't really any disagreement. I made this obviously bold
> statement as an expression of my concerns with relying on end to end as
> the sole mechanism for privacy. In the end, we did agree that it should
> stay.
>  
	Wunderbar! Thanks for the clarification.

> > 
> >      >   o It doesn't work with NAT and firewalls
> > 
> >      That's a risk you take. I refuse to dumb down protocols for
> >      everybody
> >      just because some people are behind firewalls and NATs. There are
> >      lots
> >      of protocols or features that you can't use behind a firewall or
> >      NAT.
> > 
> >      One could consider a SIP proxy as an ALG and a firewall element,
> >      could one not? If you agree with this then it does work for NAT
> >      and firewalls.
> 
> No, this does *not* work. The body of the SIP message is encrypted with
> the public key of the target in the To field. This is therefore not
> readable by anyone except the recipient, which means that NAT,
> co-located with a proxy or otherwise, can't see it. Normal proxies don't
> need to see it, but an ALG does. Thats why SIP doesn't work through NATs
> when encrypted. But, hop by hop does work, since the proxy can decrypt
> the message.
> 
	I definitely agree if we are talking about a NAT-PT/ALG that is
trying to be a SIP firewall/proxy as well.

	Do you agree that a NAT that is not concerned with the port i.e. a
simple NAT/firewall for security, denial of service, etc.. would have a
problem with this method if indeed there was a protected SIP proxy behind
the NAT/firewall?

	Sorry again - there are so many forms of NAT/proxy/firewall and
different uses for each. 

> Anyway, I believe there are uses for e2e encryption. Like everything
> else in engineering, its a tradeoff. Go for top notch security, and you
> get no services. 
> 
	While I might be mis-interpreting something, it appears to me that
you can have just as many services with e2e encryption; however, these
services must be provided by the hosts and network entitities that are
supposed to provide and protect the services. Services are worth money and
providers distinguish themselves with services. 

	SIP e2e encryption allows services to be protected at the
application level while allowing certain proxy functions that are necessary
for hop by hop routing through say a public infrastructure.

	You can't do that with other standardized session signaling
protocols that I'm aware of.

	Anyone, I'm pretty much done with this thread. Don't want to waste
people's time including my own. I hope we can agree on the value of the
functionality.


	Thanks for the clarifications, etc.. 

	Glenn


> -Jonathan R.
> 
> 
> 
> 
> -- 
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

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charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2651.65">
<TITLE>RE: [SIP] Minutes of SIP Security task force in Adelaide</TITLE>
</HEAD>
<BODY>
<BR>
<BR>
<UL>
<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Below </FONT>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; Glenn Morrow wrote:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&gt; Jonathan Rosenberg opened the discussion by suggesting =
that:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&gt; - SIP encryption (not authentication) should be deprecated,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
because:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I =
strongly disagree with this. SIP encryption serves a number of</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
useful</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
purposes. It hides information, for example, about Contact,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Subject,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Organization, Retry-After, media types, media IP addresses and</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
meeting</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
times (in SDP), SDP k=3D keys, etc. All of these are irrelevant =
for</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
proxy</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
routing behavior, but highly sensitive.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I =
agree with the disagreement.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">There isn't really any disagreement. I =
made this obviously bold</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">statement as an expression of my =
concerns with relying on end to end as</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">the sole mechanism for privacy. In =
the end, we did agree that it should</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">stay.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial"></FONT>&nbsp;
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Wunderbar! Thanks =
for the clarification.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&gt;&nbsp;&nbsp; o It doesn't work with NAT and firewalls</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
That's a risk you take. I refuse to dumb down protocols for</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
everybody</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
just because some people are behind firewalls and NATs. There =
are</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
lots</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; of =
protocols or features that you can't use behind a firewall or</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
NAT.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
One could consider a SIP proxy as an ALG and a firewall element,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
could one not? If you agree with this then it does work for NAT</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
and firewalls.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">No, this does *not* work. The body of =
the SIP message is encrypted with</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">the public key of the target in the =
To field. This is therefore not</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">readable by anyone except the =
recipient, which means that NAT,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">co-located with a proxy or otherwise, =
can't see it. Normal proxies don't</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">need to see it, but an ALG does. =
Thats why SIP doesn't work through NATs</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">when encrypted. But, hop by hop does =
work, since the proxy can decrypt</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">the message.</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">I definitely agree =
if we are talking about a NAT-PT/ALG that is trying to be a SIP =
firewall/proxy as well.</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Do you agree that a =
NAT that is not concerned with the port i.e. a simple NAT/firewall for =
security, denial of service, etc.. would have a problem with this =
method if indeed there was a protected SIP proxy behind the =
NAT/firewall?</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Sorry again - there =
are so many forms of NAT/proxy/firewall and different uses for =
each.</FONT>=20
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Anyway, I believe there are uses for =
e2e encryption. Like everything</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">else in engineering, its a tradeoff. =
Go for top notch security, and you</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">get no services. </FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">While I might be =
mis-interpreting something, it appears to me that you can have just as =
many services with e2e encryption; however, these services must be =
provided by the hosts and network entitities that are supposed to =
provide and protect the services. Services are worth money and =
providers distinguish themselves with services. </FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">SIP e2e encryption =
allows services to be protected at the application level while allowing =
certain proxy functions that are necessary for hop by hop routing =
through say a public infrastructure.</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">You can't do that =
with other standardized session signaling protocols that I'm aware =
of.</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Anyone, I'm pretty =
much done with this thread. Don't want to waste people's time including =
my own. I hope we can agree on the value of the =
functionality.</FONT></P>
<BR>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Thanks for the =
clarifications, etc.. </FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Glenn</FONT>
</P>
<BR>

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

<P><FONT SIZE=3D2 FACE=3D"Arial">-- </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Jonathan D. =
Rosenberg&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
72 Eagle Rock Ave.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">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 =
FACE=3D"Arial">dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; East Hanover, NJ 07936</FONT>
<BR><FONT SIZE=3D2 =
FACE=3D"Arial">jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; FAX:&nbsp;&nbsp; (732) 741-4778</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://www.cs.columbia.edu/~jdrosen" =
TARGET=3D"_blank">http://www.cs.columbia.edu/~jdrosen</A></FONT></U><FON=
T SIZE=3D2 =
FACE=3D"Arial">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: =
(732) 741-7244</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://www.dynamicsoft.com" =
TARGET=3D"_blank">http://www.dynamicsoft.com</A></FONT></U>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">_______________________________________=
________</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">SIP mailing list</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">SIP@lists.bell-labs.com</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
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From sip-admin@lists.bell-labs.com  Tue May 16 16:51:39 2000
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From: David Daiker <ddaiker@cisco.com>
Message-Id: <200005162050.QAA19770@blanc.cisco.com>
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
To: jdrosen@dynamicsoft.com (Jonathan Rosenberg)
Date: Tue, 16 May 2000 16:50:23 -0400 (EDT)
Cc: mat@cisco.com (Michael Thomas),
        giuseppe.ricagni@italtel.it (Giuseppe Ricagni),
        James.Kempf@Eng.Sun.COM (James Kempf), sip@lists.bell-labs.com
In-Reply-To: <392080EB.5047280@dynamicsoft.com> from "Jonathan Rosenberg" at May 15, 2000 06:57:47 PM
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> >  >    * some people claim the "proxy" should also perform some kind of admission
> >  >      control for the calls. The proxy in the home domain has no information on the
> >  >      visited domain
> > 
> >    This is true of the foreign domain too. How does
> >    any proxy have knowledge of network topology?
> 
> Call admission control in proxies >>will never work<<. As Mike suggests,
> this belongs in the transport layer, not the signaling layer.
> 
> -Jonathan R.
> -- 

Doesn't a proxy do this everytime it returns a 401 or 407 response
to an INVITE ?

Isn't PacketCable DCS doing this also ?

How about h.323 ? are you suggesting that if I am visiting China,
and make calls from a h.323 endpoint I can forward admission and location
requests to my favorite gatekeeper back in the States ?

I'm obviously missing something...

david




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David Daiker wrote:
> 
> > >  >    * some people claim the "proxy" should also perform some kind of admission
> > >  >      control for the calls. The proxy in the home domain has no information on the
> > >  >      visited domain
> > >
> > >    This is true of the foreign domain too. How does
> > >    any proxy have knowledge of network topology?
> >
> > Call admission control in proxies >>will never work<<. As Mike suggests,
> > this belongs in the transport layer, not the signaling layer.
> >
> > -Jonathan R.
> > --
> 
> Doesn't a proxy do this everytime it returns a 401 or 407 response
> to an INVITE ?

Um, no. Proxies generate 407s to authenticate users to SIP resources,
not perform admission control based on QoS for the media path.

> 
> Isn't PacketCable DCS doing this also ?

No. The admission control for RSVP is still done in the media path. It
is only policy based authorization that is done in the proxy, and this
authorization is passed to the gate performing QoS admission control.

> 
> How about h.323 ? are you suggesting that if I am visiting China,
> and make calls from a h.323 endpoint I can forward admission and location
> requests to my favorite gatekeeper back in the States ?

No. I am saying no proxy can usefully make admission control decisions
based on bandwidth, QoS, or media path resource availability. This holds
for the proxy in China and in the US. They can, of course, make
"admission control" decisions about access to resources they actually
provide, such as SIP features.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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Jonathan, Michael:
> >
> > Giuseppe Ricagni writes:
> >  >    * possible optimizations (e.g.: signalling for a basic call to an Australian
> >  >      fixed network number from an Italian mobile roaming to Australia could be
> >  >      processed locally, without any "tromboning")
> >
> >    If during call setup there needs to be a round
> >    trip to the home network, it pretty much negates
> >    any advantage of proxying the call locally.
>
> Please note that this service (optimized dial plans based on roaming) is
> still possible even if I go directly to the home network. All I need to
> do is indicate to my home proxy where I am, so it knows what dial plan
> to apply. In fact, dial plan administration is, in general, a service
> that can be outsourced to any ASP (of course, only local ASPs may wish
> to provide it, but the point is anyone can).
>

I was just talking about the PDD (post dial delay) for a call that can
be processed
locally, without any kind of message sent to the home network.

Morever, if we start from the assumption (for the above mentioned
reasons) that all SIP
signalling goes through at least one proxy, the advantages of having a
local proxy is
relevant for example in case of (SIP, hard) handovers (I' m always
referring to Henning' s approach to SIP mobility). The reduced RTT
results in a much smaller number of lost packets.

Giuseppe

--
----------------------------------------------------------
Giuseppe RICAGNI
System Architecture and Product Planning
Broadband Networks
Italtel
via Reiss Romoli - C10
20019 Castelletto di Settimo Milanese (MILANO)
ITALY

mailto:giuseppe.ricagni@italtel.it
----------------------------------------------------------



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From sip-admin@lists.bell-labs.com  Tue May 16 18:50:34 2000
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From: James Kempf <James.Kempf@Eng.Sun.COM>
Message-Id: <200005162222.PAA05644@nasnfs.eng.sun.com>
Date: Tue, 16 May 2000 15:25:02 -0700
To: "Michael Thomas" <mat@cisco.com>, <James.Kempf@Eng.Sun.COM>
Cc: "Giuseppe Ricagni" <giuseppe.ricagni@italtel.it>,
        <sip@lists.bell-labs.com>, "Michael Thomas" <mat@cisco.com>,
        "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
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Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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>James Kempf writes:
> > I don't think the analogy between SIP and HTTP is valid. I see it more
> > as being between SIP and routing. Suppose I would have to go all the
> > way back to my home network to get my packets routed. 
>
>   This is a pretty dubious analogy. We're talking
>   about proxy-routed signaling traffic and that
>   alone. You're making it sound like the media
>   would do dog-leg routing through the home
>   network which is absurd.
>

No, that's not what I meant. Thhe point I was trying to make is that 
I don't think people will be very happy if it takes them
substantially longer to make a telephone connection remotely than it
does at home. It can take somewhat longer (as it does today in the
circuit switched world) but not *substantially* longer. I don't have
the numbers for intercontinental routing delays in my head but I can
believe they must be more than routing delays to a local ISP. Whether
those delays will, in fact, be substantial enough to merit having
local proxies is a matter that can be solved by engineering analysis
(and not political posturing, as this thread seems to have degenerated into).

> > Even if you are willing to grant the analogy, HTTP requests nowadays often
> > don't get routed to the server that somebody set up as their web site.
> > If your ISP is concerned about service, they may have set up an Inktomi
> > cache, and if the publisher is concerned, they may have contracted with
> > Akami to do the same. So the world is not so simple as "let's make it
> > free and easy for anybody to set up service" on the Web anymore. 
>
>   You're assuming that every thing is going to
>   be a gigantic commercial service. I disagree. 
>   It's still perfectly possible for me to run a
>   web server at home, and in fact I do. We
>   haven't engineered walls around that
>   possibility. Requiring that I use a foreign
>   proxy does, or at least thinks it does.
>

Your web server at home is leveraging off the fact that a company,
probably AT&T back when they had a monopoly, strung a wire to your
house and installed switching equipment at the local office. They made the infrastructue
investment years ago that now allows you and your ISP to offer service
really cheaply. You may be limited as to the bandwidth you get, due
to the fact that your current local telephone provider is not willing
to make the infrastructure investment to upgrade to fiber.

The UK government just auctioned off UMTS licenses for something like $25B
(as in 1 with 9 zeros behind it). Anyone who is not able to come up
with that kind of cash will have a hard time offering wireless 
telephony service. It will take years to amortize that kind of investment.
Of course, there is always the unlicensed spectrum and I'm sure there
will be players like Metricom who figure out creative ways to utilize it.

>   Also: you're talking about transparent caching,
>   not proxying in the web case. I've not heard
>   of any ISP that *require* that I go through 
>   their HTTP proxies.
>
> > You can
> > set up your service, but unless you have a concerned ISP or the money
> > to pay Akami, the latency in using your service may cause people to
> > go elsewhere.
>
>   This analogy is fatally flawed. Cache's are 
>   great for the mountains of .gif's. There aren't
>   similar considerations for SIP. Also: even
>   if you're talking about co-lo servers
>   distributed geographically, the site owner
>   still "owns" the site. What you're talking
>   about here is something run by their
>   *competition* on your subscriber's behalf.
>
>   Yuck.
> 

Cellular companies already have romaing agreements. There's nothing new about
it. 

With regard to caches v.s. proxies, the only point I was trying to make was
that customers want fast service. Whether that service is web page access
or connecting for a telephone call. It therefore makes sense to engineer
such that the service provider can provide that service, whether it requires
transparent caches or local proxies.

> > I know this isn't going to be a popular viewpoint, but I think we need
> > to be realistic about the design. SIP already contains support for
> > telling an incoming caller that a client has moved, I can't see the
> > harm in using a remote proxy, given the performance concerns.
>
>   I think we have to demonstrate that there
>   are legitimate performance concerns before we
>   solve for them.
>

Agreed.

>   Sanity check: we're talking about a
>   hypothetical case where the SIP UA is
>   in the handset, right? We're not talking
>   about the case where you're terminating SIP
>   at the next logical hop up (BTS, MSC) for
>   legacy, right? I'm talking about the former.
>   In the latter, its debatable which proxy
>   should do routing.
>

I was talking about both. The legacy situtation will be important for
some time, particularly in the North American case where the 2G and
3G spectrum allocations overlap. But I think the core network design
should be there for smooth transition to the SIP based handset.

			jak



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James Kempf writes:
 > No, that's not what I meant. Thhe point I was trying to make is that 
 > I don't think people will be very happy if it takes them
 > substantially longer to make a telephone connection remotely than it
 > does at home. It can take somewhat longer (as it does today in the
 > circuit switched world) but not *substantially* longer. 

   Question: isn't it the case that the roamed 
   system needs to do a database dip on the
   home system to get their subscriber
   information? If so, even in the current
   situation, you have a round trip to the home
   system. This is why I'm extremely dubious that
   there is any substantial difference in call
   setup time. The same would be true in the scenario
   you propose: in order to proxy my service onto
   the roamed proxy, it would need to do some
   database dip back to my home proxy to find out
   my features. If I routed it through my home
   proxy, it looks like it would actually be
   *faster* because I'd only need to go through
   my home proxy rather than
   foreign-proxy->database-dip-at-home->routing.

 > I don't have
 > the numbers for intercontinental routing delays in my head but I can
 > believe they must be more than routing delays to a local ISP. 

   Speed of light on fiber is about 20ms/1000miles 
   as I recall.

 > Your web server at home is leveraging off the fact that a company,
 > probably AT&T back when they had a monopoly, strung a wire to your
 > house and installed switching equipment at the local office. They made the infrastructue
 > investment years ago that now allows you and your ISP to offer service
 > really cheaply.

   And I bow before New Jersey twice daily
   to give thanks to Bell above. 

 > The UK government just auctioned off UMTS licenses for something like $25B
 > (as in 1 with 9 zeros behind it). Anyone who is not able to come up
 > with that kind of cash will have a hard time offering wireless 
 > telephony service. It will take years to amortize that kind of investment.
 > Of course, there is always the unlicensed spectrum and I'm sure there
 > will be players like Metricom who figure out creative ways to utilize it.

   Oh, please. Let's keep track of where the actual
   investment is here: it's the air resources and 
   the raw bandwidth it enables. There's an
   incremental delta for the switching equipment,
   but that's by and large the same equipment as
   landline. This doesn't seem very hard to me:
   if it's expensive to put bits in the air, then
   you charge for it. Let's not get taken down the
   path that the only viable business model for a
   bandwidth provider is to have monopoly access
   to the services I can access.

   That's the same mentality that didn't invent the
   Internet. The same Internet which dramatically
   demonstrates that there are other business
   models which can turn a tidy profit being a
   boring unintelligent network operator. If there
   are good engineering reasons, fine. But
   please lets not get sucked into bad engineering
   design driven by old world business models.

 > Cellular companies already have romaing agreements. There's nothing new about
 > it. 

   I know they do, and my understanding is that
   feature transparency is one thing that suffers
   big time because of their current model. 
   Keeping my features back where I get my 
   features eliminates that feature synch
   problem at its root. That is impossible in
   the circuit switched world, but almost trivial
   in the SIP world.

 > >   I think we have to demonstrate that there
 > >   are legitimate performance concerns before we
 > >   solve for them.
 > >
 > 
 > Agreed.

   I think what might be useful rather than another
   round of impassioned polemics here is to see
   some message flows, paying particular attention
   to any hidden assumptions about what the home
   network needs to provide.

		 Mike


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From sip-admin@lists.bell-labs.com  Wed May 17 00:24:37 2000
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Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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Giuseppe Ricagni wrote:
> 
> Jonathan, Michael:
> > >
> > > Giuseppe Ricagni writes:
> > >  >    * possible optimizations (e.g.: signalling for a basic call to an Australian
> > >  >      fixed network number from an Italian mobile roaming to Australia could be
> > >  >      processed locally, without any "tromboning")
> > >
> > >    If during call setup there needs to be a round
> > >    trip to the home network, it pretty much negates
> > >    any advantage of proxying the call locally.
> >
> > Please note that this service (optimized dial plans based on roaming) is
> > still possible even if I go directly to the home network. All I need to
> > do is indicate to my home proxy where I am, so it knows what dial plan
> > to apply. In fact, dial plan administration is, in general, a service
> > that can be outsourced to any ASP (of course, only local ASPs may wish
> > to provide it, but the point is anyone can).
> >
> 
> I was just talking about the PDD (post dial delay) for a call that can
> be processed
> locally, without any kind of message sent to the home network.

I think the RTT for going to the home network is going to be a drop in
the bucket compared to the typical PDDs in wireless today. The RTT for
going to the home network is going to have to be reasonable to carry
voice anyway, so we are talking a few hundred milliseconds, worst case?
It takes my cell phone 5 seconds for PDD now....


> 
> Morever, if we start from the assumption (for the above mentioned
> reasons) that all SIP
> signalling goes through at least one proxy, the advantages of having a
> local proxy is
> relevant for example in case of (SIP, hard) handovers (I' m always
> referring to Henning' s approach to SIP mobility). The reduced RTT
> results in a much smaller number of lost packets.

How is that? Its a dangerous game to make strong assumptions on loss
models like this.

In any case, I think that optimizing an architecture based on
assumptions about loss distributions and latency is somewhat
short-sighted. Lousy network quality is going to kill you on the
compressed voice traffic long before the signaling gets really blown
away. Even from a human factors perspective, most people will tolerate
long PDDs but not high voice packet loss rates. If the signaling loss is
really bothering you, deploy good networks with diffserv giving great
service to port 5060. Eventually these network performance problems will
get solved. But, fundamental architectural decisions don't get changed.
The ability to choose anyone as my applications provider, and not suffer
horrible voice quality (since the audio always goes direct) is such a
huge win for competition and service innovation. Saying no to that to
save 50ms on signaling latency seems such a shame. This was my point
about web. If I had to *rely* on my ISP replicating the content in order
to fetch it, there never would have been a web. It is only because
anyone, anywhere on the Internet could set up shop, and anyone, anywhere
could access it, that we had the huge growth in this service. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Wed May 17 01:42:48 2000
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "James Kempf" <James.Kempf@Eng.Sun.COM>
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Subject: RE: [SIP] Minutes of SIP Security task force in Adelaide
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But what if your speed-dial list and your find-others scripts and your automatically-included features like recording and transcription are activated by your home proxy?

Personally, I plan to use the custom services I've built around my environment even when I'm roaming, and I plan to build many of those services into my home proxy.

--
Dean

> -----Original Message-----
> From: sip-admin@lists.bell-labs.com
> [mailto:sip-admin@lists.bell-labs.com]On Behalf Of James Kempf
> Sent: Saturday, May 13, 2000 9:22 PM
> To: James.Kempf@Eng.Sun.COM; mat@cisco.com
> Cc: sip@lists.bell-labs.com; jdrosen@dynamicsoft.com
> Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
> 
> 
> 
> > > Certainly in the case of cellular, it will be the case that the
> > > proxy will be changing. When the mobile terminal roams outside
> > > of it's cellular provider's domain, the new proxy needs to be 
> authenticated
> > > with the mobile terminal, and vice versa. 
> >
> >   That sort of pre-supposes that the proxy the
> >   handset contacts when it's roamed isn't its
> >   home proxy. I'm not sure I see why that
> >   wouldn't still be the case. I hope people
> >   aren't getting buffaloed into some garden
> >   walls kind of mentality where the service
> >   provider gets to keep their stranglehold on
> >   services.
> >
> 
> It's not the case because if I have my cellular account with a provider
> in California and I travel to Australia, I don't want to have to 
> go all the way back to the SIP proxy with my provider in California
> to have to make a call.
> 
> 
> 		jak
> 
> 
> 
> _______________________________________________
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: "IETF SIP" <sip@lists.bell-labs.com>
Subject: RE: [SIP] Minutes of SIP Security task force in Adelaide
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Jonathan wrote:
> No. I am saying no proxy can usefully make admission control decisions
> based on bandwidth, QoS, or media path resource availability. This holds
> for the proxy in China and in the US. They can, of course, make
> "admission control" decisions about access to resources they actually
> provide, such as SIP features.

Actually, I think there might be an exception. Let's say we have a large network that mixes several DiffServe classes -- for simplicity, let's say "best effort" and "priority" clases. Assume that there is normally a relatively high ratio of best-effort to priority traffic in the network.

One might a heuristic for traffic engineering that works like this:

Assume a network monitoring system that can provide both a rough estimate of the ration of best-effort to priority traffic on the backbone and on specific links. Feed this data into access proxies.

Define a minimum ratio for both of those metrics, say 2-1.  A proxy could then provide a "network health guidance" effect for edge devices.

Sure, we can't KEEP an unfriendly app from jumping in during a peak congestion period, but we could certainly ADVISE friendly apps that it woudn't be a good idea to send more traffic right now.

Or one could just put the QoS function under control of the proxy, which activates the last clause in Jonathan's  argument.

--
Dean

--
Dean
ÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÒ ùšŠX§‚X¬µ"þX¬¶ÏÛzYÿ•¦ìýÊ&†ÛiÿÿåŠËlý·¥—ùZnÏÜ¢oæj)fjåŠËbú?


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Date: Wed, 17 May 2000 08:26:55 +0200
From: Stephen Terrill <stephen.terrill@ericsson.com>
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
To: James.Kempf@Eng.Sun.COM
Cc: Michael Thomas <mat@cisco.com>,
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Hi James,

I think that the emphansis on saving the trombing of the user signalling is much overweighted.  Firstly, the "singlling" transport in these networks is not limited to the 64kbs singnalling channels used in the PSTN today.  Secondly, even today, it is not uncommon for two operators in the same country (ie a european country) to interconnect to each other by going through the US due to commercial reasons (ie a call from one phone to another phone just 30 meeters away can cross the atlantic twice).  What I am saying as that this happens today, and that is both user signalling and payload, and customer usually aren´t aware of it.

As such, I do tend to agree with Jonathan and Mike that the advantages of taking the call/session signalling to the "home" proxy outweigh this small and dissappearing disadvantage.

Regards,

..//steve

James Kempf wrote:

> >James Kempf writes:
> > > I don't think the analogy between SIP and HTTP is valid. I see it more
> > > as being between SIP and routing. Suppose I would have to go all the
> > > way back to my home network to get my packets routed.
> >
> >   This is a pretty dubious analogy. We're talking
> >   about proxy-routed signaling traffic and that
> >   alone. You're making it sound like the media
> >   would do dog-leg routing through the home
> >   network which is absurd.
> >
>
> No, that's not what I meant. Thhe point I was trying to make is that
> I don't think people will be very happy if it takes them
> substantially longer to make a telephone connection remotely than it
> does at home. It can take somewhat longer (as it does today in the
> circuit switched world) but not *substantially* longer. I don't have
> the numbers for intercontinental routing delays in my head but I can
> believe they must be more than routing delays to a local ISP. Whether
> those delays will, in fact, be substantial enough to merit having
> local proxies is a matter that can be solved by engineering analysis
> (and not political posturing, as this thread seems to have degenerated into).
>
> > > Even if you are willing to grant the analogy, HTTP requests nowadays often
> > > don't get routed to the server that somebody set up as their web site.
> > > If your ISP is concerned about service, they may have set up an Inktomi
> > > cache, and if the publisher is concerned, they may have contracted with
> > > Akami to do the same. So the world is not so simple as "let's make it
> > > free and easy for anybody to set up service" on the Web anymore.
> >
> >   You're assuming that every thing is going to
> >   be a gigantic commercial service. I disagree.
> >   It's still perfectly possible for me to run a
> >   web server at home, and in fact I do. We
> >   haven't engineered walls around that
> >   possibility. Requiring that I use a foreign
> >   proxy does, or at least thinks it does.
> >
>
> Your web server at home is leveraging off the fact that a company,
> probably AT&T back when they had a monopoly, strung a wire to your
> house and installed switching equipment at the local office. They made the infrastructue
> investment years ago that now allows you and your ISP to offer service
> really cheaply. You may be limited as to the bandwidth you get, due
> to the fact that your current local telephone provider is not willing
> to make the infrastructure investment to upgrade to fiber.
>
> The UK government just auctioned off UMTS licenses for something like $25B
> (as in 1 with 9 zeros behind it). Anyone who is not able to come up
> with that kind of cash will have a hard time offering wireless
> telephony service. It will take years to amortize that kind of investment.
> Of course, there is always the unlicensed spectrum and I'm sure there
> will be players like Metricom who figure out creative ways to utilize it.
>
> >   Also: you're talking about transparent caching,
> >   not proxying in the web case. I've not heard
> >   of any ISP that *require* that I go through
> >   their HTTP proxies.
> >
> > > You can
> > > set up your service, but unless you have a concerned ISP or the money
> > > to pay Akami, the latency in using your service may cause people to
> > > go elsewhere.
> >
> >   This analogy is fatally flawed. Cache's are
> >   great for the mountains of .gif's. There aren't
> >   similar considerations for SIP. Also: even
> >   if you're talking about co-lo servers
> >   distributed geographically, the site owner
> >   still "owns" the site. What you're talking
> >   about here is something run by their
> >   *competition* on your subscriber's behalf.
> >
> >   Yuck.
> >
>
> Cellular companies already have romaing agreements. There's nothing new about
> it.
>
> With regard to caches v.s. proxies, the only point I was trying to make was
> that customers want fast service. Whether that service is web page access
> or connecting for a telephone call. It therefore makes sense to engineer
> such that the service provider can provide that service, whether it requires
> transparent caches or local proxies.
>
> > > I know this isn't going to be a popular viewpoint, but I think we need
> > > to be realistic about the design. SIP already contains support for
> > > telling an incoming caller that a client has moved, I can't see the
> > > harm in using a remote proxy, given the performance concerns.
> >
> >   I think we have to demonstrate that there
> >   are legitimate performance concerns before we
> >   solve for them.
> >
>
> Agreed.
>
> >   Sanity check: we're talking about a
> >   hypothetical case where the SIP UA is
> >   in the handset, right? We're not talking
> >   about the case where you're terminating SIP
> >   at the next logical hop up (BTS, MSC) for
> >   legacy, right? I'm talking about the former.
> >   In the latter, its debatable which proxy
> >   should do routing.
> >
>
> I was talking about both. The legacy situtation will be important for
> some time, particularly in the North American case where the 2G and
> 3G spectrum allocations overlap. But I think the core network design
> should be there for smooth transition to the SIP based handset.
>
>                         jak
>
> _______________________________________________
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> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip



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From sip-admin@lists.bell-labs.com  Wed May 17 03:49:24 2000
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When a UAS issues a 400+ response to a re-INVITE, is it signaling that a)
the entire call has been torn down  or b) that only the re-INVITE failed but
the call is still active? I couldn't find anywhere this has been clarified.

Both scenarios have their uses and pitfalls.

The former case allows no room for correction/negotiation of the failed
message component. Problems also arise if one side or one of the
intermediate proxies assumes that the call has not been torn down.

Likewise, if the latter alternative is true, then it is easily conceivable
that a UAC might incorrectly issue an ACK instead of a BYE to the failed
re-INVITE. Now the UAC has torn down the call but the UAS has not. 

I guess you could say that this is simply another reason why the SIP Session
Timer is a good idea. 

Cheers,

Robert.


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From sip-admin@lists.bell-labs.com  Wed May 17 04:06:28 2000
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Subject: Re: [SIP] Error response to re-INVITE.
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Hi,

The current understanding is that if a RE-INVITE fails - its valid only for
the reinvite and not for the entire call.
i.e. the previous call parameters (and hence the call) are still active.


robert> Likewise, if the latter alternative is true, then it is easily
conceivable
robert>  that a UAC might incorrectly issue an ACK instead of a BYE to the
failed
robert>  re-INVITE. Now the UAC has torn down the call but the UAS has not.

I dont follow you here. Why is this a problem ? the usual
INVITE-FinalResponse-ACK happens here. Both parties know the RE-INVITE
failed and hence the old call proceeds.
The RE-INVITE issuing party as it is needs to send an ACK to the final
response from the other party.
And a BYE  Iguess will be sent only if the calling party decides that if
the re-INVITE fails, then he is not interested in carrying on with the old
parameters.

Regds
Arjun

--
Arjun Roychowdhury @ Hughes Software Systems






"Fairlie-Cuninghame, Robert" <rfairlie@nuera.com> on 05/17/2000 01:19:17 PM

To:   IETF SIP <sip@lists.bell-labs.com>
cc:

Subject:  [SIP] Error response to re-INVITE.





When a UAS issues a 400+ response to a re-INVITE, is it signaling that a)
the entire call has been torn down  or b) that only the re-INVITE failed
but
the call is still active? I couldn't find anywhere this has been clarified.

Both scenarios have their uses and pitfalls.

The former case allows no room for correction/negotiation of the failed
message component. Problems also arise if one side or one of the
intermediate proxies assumes that the call has not been torn down.

Likewise, if the latter alternative is true, then it is easily conceivable
that a UAC might incorrectly issue an ACK instead of a BYE to the failed
re-INVITE. Now the UAC has torn down the call but the UAS has not.

I guess you could say that this is simply another reason why the SIP
Session
Timer is a good idea.

Cheers,

Robert.


_______________________________________________
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SIP@lists.bell-labs.com
http://lists.bell-labs.com/mailman/listinfo/sip






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>
>   Question: isn't it the case that the roamed 
>   system needs to do a database dip on the
>   home system to get their subscriber
>   information? If so, even in the current
>   situation, you have a round trip to the home
>   system. This is why I'm extremely dubious that
>   there is any substantial difference in call
>   setup time. The same would be true in the scenario
>   you propose: in order to proxy my service onto
>   the roamed proxy, it would need to do some
>   database dip back to my home proxy to find out
>   my features. If I routed it through my home
>   proxy, it looks like it would actually be
>   *faster* because I'd only need to go through
>   my home proxy rather than
>   foreign-proxy->database-dip-at-home->routing.
>

With the proper security association, the subscriber profile could be forwarded
to the roamed proxy, avoiding the round trip back to the home proxy. This
actually would be a *big* saving from the current cellular networks, where
the remote VLR has to go back to the HLR every time. I've seen some research
work on this for current circuit switched, but it is crufty. It is lots
easier to do it with an all IP network, provided the proper AAA protocols
are in place.

> > I don't have
> > the numbers for intercontinental routing delays in my head but I can
> > believe they must be more than routing delays to a local ISP. 
>
>   Speed of light on fiber is about 20ms/1000miles 
>   as I recall.
>

That's transport. There's also routing. What's the queueing delay?

> > Your web server at home is leveraging off the fact that a company,
> > probably AT&T back when they had a monopoly, strung a wire to your
> > house and installed switching equipment at the local office. They made the
>infrastructue
> > investment years ago that now allows you and your ISP to offer service
> > really cheaply.
>
>   And I bow before New Jersey twice daily
>   to give thanks to Bell above. 
>
> > The UK government just auctioned off UMTS licenses for something like $25B
> > (as in 1 with 9 zeros behind it). Anyone who is not able to come up
> > with that kind of cash will have a hard time offering wireless 
> > telephony service. It will take years to amortize that kind of investment.
> > Of course, there is always the unlicensed spectrum and I'm sure there
> > will be players like Metricom who figure out creative ways to utilize it.
>
>   Oh, please. Let's keep track of where the actual
>   investment is here: it's the air resources and 
>   the raw bandwidth it enables. There's an
>   incremental delta for the switching equipment,
>   but that's by and large the same equipment as
>   landline. This doesn't seem very hard to me:
>   if it's expensive to put bits in the air, then
>   you charge for it. Let's not get taken down the
>   path that the only viable business model for a
>   bandwidth provider is to have monopoly access
>   to the services I can access.
>

It's not a monopoly, but there are very few companies in the world that are
in the position to plunk down $5B or so for a UMTS license. And governments
aren't offering many. It is partially the nature of the radio spectrum
(too many users leads to interference) and partially business politics
as usual.

>   That's the same mentality that didn't invent the
>   Internet. The same Internet which dramatically
>   demonstrates that there are other business
>   models which can turn a tidy profit being a
>   boring unintelligent network operator. If there
>   are good engineering reasons, fine. But
>   please lets not get sucked into bad engineering
>   design driven by old world business models.
>
> > Cellular companies already have romaing agreements. There's nothing new
>about
> > it. 
>
>   I know they do, and my understanding is that
>   feature transparency is one thing that suffers
>   big time because of their current model. 
>   Keeping my features back where I get my 
>   features eliminates that feature synch
>   problem at its root. That is impossible in
>   the circuit switched world, but almost trivial
>   in the SIP world.
>

If features can be easily forwarded between proxies (with the proper
authentication) then it's even easier to get your features whereever
you want them.

> > >   I think we have to demonstrate that there
> > >   are legitimate performance concerns before we
> > >   solve for them.
> > >
> > 
> > Agreed.
>
>   I think what might be useful rather than another
>   round of impassioned polemics here is to see
>   some message flows, paying particular attention
>   to any hidden assumptions about what the home
>   network needs to provide.
>

Agreed! Putting numbers on message flows would answer the question. 
>		 Mike
>
>
>_______________________________________________
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>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip




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From sip-admin@lists.bell-labs.com  Wed May 17 08:16:32 2000
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hi

I have a problem with the In-Reply-To header, 2543 bis section 6.25 p. 48:

"The In-Reply-To request header field enumerates the call-IDs that this call references or returns.
This allows automatic call distribution systems to route return calls to the originator of the first call and allows
callees to filter calls, so that only calls that return calls they have originated will be accepted. This field is not a
substitute for request authentication.

In-Reply-To =  In-Reply-To   :  1# callid"


If i can refer you to Anders email titled:

Re: [SIP] Another grammar query - Call-ID

then callid has been re-defined as 
"Basically, the field should be treated as an opaque,
uninterpreted identifier, e.g. TEXT-UTF8-TRIM in the grammar."

The problem is that if the In-Reply-To header is:
In-Reply-To: 70710@saturn.bell-tel.com, 17320@saturn.bell-tel.com

then this will resolve as a single call-id not two separate ones.

would it not be possible to say make call-id a quoted-string???
This may make little sense for the Call-ID header but it solves the problem here.
what would we then compare on though?  could escaped values get in here which would
stop the efficient byte by byte equality checking that TEXT-UTF8-TRIM provided?

cheers

gethin

-- 
Gethin Liddell
Ubiquity Software Corporation

http://www.ubiquity.net
mailto:gethin@ubiquity.net


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From sip-admin@lists.bell-labs.com  Wed May 17 08:37:13 2000
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From: Giuseppe Ricagni <giuseppe.ricagni@italtel.it>
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Stephen,

Stephen Terrill wrote:

> Hi James,
>
> I think that the emphansis on saving the trombing of the user signalling is much overweighted.  Firstly, the "singlling" transport in these networks is not limited to the 64kbs singnalling channels used in the PSTN today.

Higher bandwidth doesn' t necessarily mean lower latency..... Moreover
you don' t normally have *just one* 64Kb/s link. In this
specific case, the tight delay requirements on SS7 networks, might well
result in lower RTTs for signalling messages.

> Secondly, even today, it is not uncommon for two operators in the same country (ie a european country) to interconnect to each other by going through the US due to commercial reasons (ie a call from one phone to another phone just 30 meeters away can cross the atlantic twice).  What I am saying as that this happens today, and that is both user signalling and payload, and customer usually aren´t aware of it.
>

not saying that signalling should *always* be processed locally: as you
know this is just impossible, especially when user profile data is
required to process the call.

What we are claiming here is that local procesing should be employed as
an optimization, in all those cases where it is actually possible,
without any (pre-call) data exchange with the home domain.


> As such, I do tend to agree with Jonathan and Mike that the advantages of taking the call/session signalling to the "home" proxy outweigh this small and dissappearing disadvantage.

I do tend to disagree... Why not applying an optimization (only in those
cases where it can be applied, obviously) if it's easy, it has basically
no disadvantages, and it can be even much more helpful in other contexts
(seen my other posting ?)

Cheers
Giuseppe
P.S.
BTW: you will remember that  “Optimised call routing” is a requirement
for 3G networks (it does also refer to signalling)....

>
>
> Regards,
>
> ..//steve
>
> James Kempf wrote:
>
> > >James Kempf writes:
> > > > I don't think the analogy between SIP and HTTP is valid. I see it more
> > > > as being between SIP and routing. Suppose I would have to go all the
> > > > way back to my home network to get my packets routed.
> > >
> > >   This is a pretty dubious analogy. We're talking
> > >   about proxy-routed signaling traffic and that
> > >   alone. You're making it sound like the media
> > >   would do dog-leg routing through the home
> > >   network which is absurd.
> > >
> >
> > No, that's not what I meant. Thhe point I was trying to make is that
> > I don't think people will be very happy if it takes them
> > substantially longer to make a telephone connection remotely than it
> > does at home. It can take somewhat longer (as it does today in the
> > circuit switched world) but not *substantially* longer. I don't have
> > the numbers for intercontinental routing delays in my head but I can
> > believe they must be more than routing delays to a local ISP. Whether
> > those delays will, in fact, be substantial enough to merit having
> > local proxies is a matter that can be solved by engineering analysis
> > (and not political posturing, as this thread seems to have degenerated into).
> >
> > > > Even if you are willing to grant the analogy, HTTP requests nowadays often
> > > > don't get routed to the server that somebody set up as their web site.
> > > > If your ISP is concerned about service, they may have set up an Inktomi
> > > > cache, and if the publisher is concerned, they may have contracted with
> > > > Akami to do the same. So the world is not so simple as "let's make it
> > > > free and easy for anybody to set up service" on the Web anymore.
> > >
> > >   You're assuming that every thing is going to
> > >   be a gigantic commercial service. I disagree.
> > >   It's still perfectly possible for me to run a
> > >   web server at home, and in fact I do. We
> > >   haven't engineered walls around that
> > >   possibility. Requiring that I use a foreign
> > >   proxy does, or at least thinks it does.
> > >
> >
> > Your web server at home is leveraging off the fact that a company,
> > probably AT&T back when they had a monopoly, strung a wire to your
> > house and installed switching equipment at the local office. They made the infrastructue
> > investment years ago that now allows you and your ISP to offer service
> > really cheaply. You may be limited as to the bandwidth you get, due
> > to the fact that your current local telephone provider is not willing
> > to make the infrastructure investment to upgrade to fiber.
> >
> > The UK government just auctioned off UMTS licenses for something like $25B
> > (as in 1 with 9 zeros behind it). Anyone who is not able to come up
> > with that kind of cash will have a hard time offering wireless
> > telephony service. It will take years to amortize that kind of investment.
> > Of course, there is always the unlicensed spectrum and I'm sure there
> > will be players like Metricom who figure out creative ways to utilize it.
> >
> > >   Also: you're talking about transparent caching,
> > >   not proxying in the web case. I've not heard
> > >   of any ISP that *require* that I go through
> > >   their HTTP proxies.
> > >
> > > > You can
> > > > set up your service, but unless you have a concerned ISP or the money
> > > > to pay Akami, the latency in using your service may cause people to
> > > > go elsewhere.
> > >
> > >   This analogy is fatally flawed. Cache's are
> > >   great for the mountains of .gif's. There aren't
> > >   similar considerations for SIP. Also: even
> > >   if you're talking about co-lo servers
> > >   distributed geographically, the site owner
> > >   still "owns" the site. What you're talking
> > >   about here is something run by their
> > >   *competition* on your subscriber's behalf.
> > >
> > >   Yuck.
> > >
> >
> > Cellular companies already have romaing agreements. There's nothing new about
> > it.
> >
> > With regard to caches v.s. proxies, the only point I was trying to make was
> > that customers want fast service. Whether that service is web page access
> > or connecting for a telephone call. It therefore makes sense to engineer
> > such that the service provider can provide that service, whether it requires
> > transparent caches or local proxies.
> >
> > > > I know this isn't going to be a popular viewpoint, but I think we need
> > > > to be realistic about the design. SIP already contains support for
> > > > telling an incoming caller that a client has moved, I can't see the
> > > > harm in using a remote proxy, given the performance concerns.
> > >
> > >   I think we have to demonstrate that there
> > >   are legitimate performance concerns before we
> > >   solve for them.
> > >
> >
> > Agreed.
> >
> > >   Sanity check: we're talking about a
> > >   hypothetical case where the SIP UA is
> > >   in the handset, right? We're not talking
> > >   about the case where you're terminating SIP
> > >   at the next logical hop up (BTS, MSC) for
> > >   legacy, right? I'm talking about the former.
> > >   In the latter, its debatable which proxy
> > >   should do routing.
> > >
> >
> > I was talking about both. The legacy situtation will be important for
> > some time, particularly in the North American case where the 2G and
> > 3G spectrum allocations overlap. But I think the core network design
> > should be there for smooth transition to the SIP based handset.
> >
> >                         jak
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
>
> _______________________________________________
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--
----------------------------------------------------------
Giuseppe RICAGNI
System Architecture and Product Planning
Broadband Networks
Italtel
via Reiss Romoli - C10
20019 Castelletto di Settimo Milanese (MILANO)
ITALY

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From sip-admin@lists.bell-labs.com  Wed May 17 08:38:46 2000
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Jonathan Rosenberg wrote:

> Giuseppe Ricagni wrote:
>

> > I was just talking about the PDD (post dial delay) for a call that can
> > be processed
> > locally, without any kind of message sent to the home network.
>
> I think the RTT for going to the home network is going to be a drop in
> the bucket compared to the typical PDDs in wireless today. The RTT for
> going to the home network is going to have to be reasonable to carry
> voice anyway, so we are talking a few hundred milliseconds, worst case?
> It takes my cell phone 5 seconds for PDD now....

Well, you might want to consider changing  mobile operator... :-))

>
> >
> > Morever, if we start from the assumption (for the above mentioned
> > reasons) that all SIP
> > signalling goes through at least one proxy, the advantages of having a
> > local proxy is
> > relevant for example in case of (SIP, hard) handovers (I' m always
> > referring to Henning' s approach to SIP mobility). The reduced RTT
> > results in a much smaller number of lost packets.
>
> How is that? Its a dangerous game to make strong assumptions on loss
> models like this.
>

I think we are talking about two different things.
I was just sayng that, according to Henning's model, to put it very simple, when you change
IP address you shuld send an Invite to your correspondent host containing your new IP
address (plus re-registering with your proxy, to allow for signalling to be sent to the new
address).

If we assume (by hipotesys) that all SIP messages must go through at least 1 proxy, and if
we have an active conversation between two Italian mobiles located in Australia, the so
called "handoff delay" is one RTT between Australia and Italy, which can be something like
300 msec or (very likely) more.
Even with a 300msec figure, you would en up losing some 15 voice samples, which is something
definetly undesirable.

OTOH, with a local proxy, you wouldn' t lose more than 1 or, at worst, two samples.

Giuseppe

>
> In any case, I think that optimizing an architecture based on
> assumptions about loss distributions and latency is somewhat
> short-sighted. Lousy network quality is going to kill you on the
> compressed voice traffic long before the signaling gets really blown
> away. Even from a human factors perspective, most people will tolerate
> long PDDs but not high voice packet loss rates. If the signaling loss is
> really bothering you, deploy good networks with diffserv giving great
> service to port 5060. Eventually these network performance problems will
> get solved. But, fundamental architectural decisions don't get changed.
> The ability to choose anyone as my applications provider, and not suffer
> horrible voice quality (since the audio always goes direct) is such a
> huge win for competition and service innovation. Saying no to that to
> save 50ms on signaling latency seems such a shame. This was my point
> about web. If I had to *rely* on my ISP replicating the content in order
> to fetch it, there never would have been a web. It is only because
> anyone, anywhere on the Internet could set up shop, and anyone, anywhere
> could access it, that we had the huge growth in this service.
>
> -Jonathan R.
>
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com

--
----------------------------------------------------------
Giuseppe RICAGNI
System Architecture and Product Planning
Broadband Networks
Italtel
via Reiss Romoli - C10
20019 Castelletto di Settimo Milanese (MILANO)
ITALY

mailto:giuseppe.ricagni@italtel.it
----------------------------------------------------------







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To: Giuseppe Ricagni <giuseppe.ricagni@italtel.it>
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        Michael Thomas <mat@cisco.com>, James Kempf <James.Kempf@Eng.Sun.COM>,
        sip@lists.bell-labs.com
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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There seem to be two cases:

1) The callee is local and the caller is roaming. In that case, there's
no reason to have the caller go through its remote (home) proxy for
sending SIP requests (except if that home proxy performs, for example,
authentication services).

2) The callee is roaming (independent of where the caller is). The SIP
call has no choice but to go the home proxy of the callee first, since
that's the only reference point for the caller. It has no clue that the
callee is standing right next to him/her.


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From sip-admin@lists.bell-labs.com  Wed May 17 09:27:17 2000
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The home proxy always should be within the signaling path, even though you
have roamed into another network. This is very important from a value added
service perspective. 

I have no opinion if is bad or not to force a client to signal through a
proxy in the visited network or not. There could be other reasons than
billing to do this, for example would the visited network do some logging,
or needs to route the messages through a SIP aware firewall. Note that I am
not proposing firewalls to be there, but some operators might want them... I
think that billing for media (i.e. voice) must be coupled to the transport
layer.

The reason to have the home proxy within the signaling path is to be able to
offer services that is invoked before the call is set up. One example is a
speed dial dial service that translates one request URI to something else
depending on a users preferences.

Is it correct for a UAC to add a route header to indicate to a proxy in the
visited network that it should forward the request to the home server?
Section 6.37 is a bit confusing since it says "The route request-header
field determines the route taken by a request..." I guess it should be
changed to "The route request-header field determines the route a request
will take..." since it not yet been routed this way when the field is added.

I have understood that some groups proposes a solution where the services is
downloaded to the visited network. The solution with downloaded services has
one severe problem, the fact that the execution environment must be
homogenous between the operators. This is very limiting for a operator that
want to distinguish himself and attract customers by providing creative
services, which is very important on todays competitive market. A successful
service might need to retrive information from an information system that
the users service provider has access to. This means that althouh the code
will be able to execute when it is downloaded to the visited network so will
the information that is needed not be accessible.

I don't think this would affect the call setup time in any significant way,
and users is today used to longer call setup times when the roam into
another operators network.

/Jörgen

-----Original Message-----
From: James Kempf
To: Michael Thomas; Giuseppe Ricagni
Cc: jdrosen@dynamicsoft.com; sip@lists.bell-labs.com; James Kempf; Michael
Thomas
Sent: 2000-05-16 14:05
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide


>Giuseppe Ricagni writes:
> >    * possible optimizations (e.g.: signalling for a basic call to an
>Australian
> >      fixed network number from an Italian mobile roaming to
Australia could
>be
> >      processed locally, without any "tromboning")
>
>   If during call setup there needs to be a round
>   trip to the home network, it pretty much negates
>   any advantage of proxying the call locally. 
>

When you take roaming into account, there is an advantage. Suppose the
Italian now in Australia moves. If the home proxy is used, the network
has to go back to the home to roam. If a local proxy is used, there is
reduced latency. With a local proxy, you pay the round trip cost
to the home network just once.

In addition, some people have been talking about using SIP for other
kinds of in-call signalling. If the mobile has to go all the way
back to the home proxy, there is simply too much latency.






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From sip-admin@lists.bell-labs.com  Wed May 17 09:41:57 2000
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Date: Wed, 17 May 2000 14:04:08 +0200
From: Jiri Kuthan <kuthan@fokus.gmd.de>
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Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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Bellow.

> Glenn Morrow wrote:

> 
>      No, this does *not* work. The body of the SIP message is encrypted with
>      the public key of the target in the To field. This is therefore not
>      readable by anyone except the recipient, which means that NAT,
>      co-located with a proxy or otherwise, can't see it. Normal proxies don't
>      need to see it, but an ALG does. Thats why SIP doesn't work through NATs
>      when encrypted. But, hop by hop does work, since the proxy can decrypt
>      the message.

> 
>      I definitely agree if we are talking about a NAT-PT/ALG that is trying to be a SIP firewall/proxy as well.
> 
>      Do you agree that a NAT that is not concerned with the port i.e. a simple NAT/firewall for security, denial of service, etc.. would have a problem with this method if indeed there was a protected SIP proxy behind the NAT/firewall?
> 
>      Sorry again - there are so many forms of NAT/proxy/firewall and different uses for each.
> 

There are two problems: 

1) SIP does not work with firewalls at all unless they are SIP-aware
   (probably the problem you are asking about, right?)

   Solution is straightforward: teach firewalls to speak SIP
   (reworded more generally: include ALGs); SIP-awareness can
   be either embedded in the firewalls or SIP-proxies can
   be reused for this purpose if an interface between them
   and firewalls is provided. IMHO, the latter alternative
   is preferable.

2) The second separate issue is that even if the firewalls
   are SIP-aware it doesn't help if the SIP messages are
   e2e encrypted.

   This is not a technical problem. This is network security configuration
   problem. Either your network administrator wants to eliminate traffic
   considered dangerous (e.g., SIP INVITEs with ILOVEYOU, unsolicited media) 
   centrally with firewalls  and thus has to inspect signaling (yes, this
   implies the SIP proxy administrator will know you received video from
   Mary at port X), or he has no firewalls and lets you do what you want 
   to (including privacy by encryption and receiving ILOVEYOU).

   The only technical issue here is (we seem to have consensus on it)
   a) usage of a SIP reply which says "my firewall cannot decrypt your message"
   b) go for hop-by-hop encryption (the administrator still knows about you and Mary
      but eavedroppers do not)

To answer your question: I do not see the problem with SIP-aware NATs/FWs
and HbH encryption other than your administrator seeing your signaling. If you
want to have e2e encryption of your session with Mary use your ISP w/o FWs
instead of your corporate network with them.

Jiri




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Date: Wed, 17 May 2000 10:10:05 -0400
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
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From: "Michael A. Ramalho" <mramalho@cisco.com>
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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At 12:35 AM 5/17/00 -0400, Jonathan Rosenberg wrote:
>
>
>Giuseppe Ricagni wrote:
>> 
>> Jonathan, Michael:
>> > >
>> > > Giuseppe Ricagni writes:
>> > >  >    * possible optimizations (e.g.: signalling for a basic call to 
>an Australian
>> > >  >      fixed network number from an Italian mobile roaming to 
>Australia could be
>> > >  >      processed locally, without any "tromboning")
>> > >
>> > >    If during call setup there needs to be a round
>> > >    trip to the home network, it pretty much negates
>> > >    any advantage of proxying the call locally.
>> >
>> > Please note that this service (optimized dial plans based on roaming) is
>> > still possible even if I go directly to the home network. All I need to
>> > do is indicate to my home proxy where I am, so it knows what dial plan
>> > to apply. In fact, dial plan administration is, in general, a service
>> > that can be outsourced to any ASP (of course, only local ASPs may wish
>> > to provide it, but the point is anyone can).
>> >
>> 
>> I was just talking about the PDD (post dial delay) for a call that can
>> be processed
>> locally, without any kind of message sent to the home network.
>
>I think the RTT for going to the home network is going to be a drop in
>the bucket compared to the typical PDDs in wireless today. The RTT for
>going to the home network is going to have to be reasonable to carry
>voice anyway, so we are talking a few hundred milliseconds, worst case?
>It takes my cell phone 5 seconds for PDD now....

FWIW - I would love only 5 seconds!

My home office phone (Bell Atlantic) is often forwarded to my AT&T Wireless
phone. However, I'm usually in a Comcast serving area (i.e., roaming).

When I am at home when someone calls my forwarded home office
phone ... it pings (to let me know a call has been forwarded) ... then I
wait .... and wait some more ..... usually 6 to 8 seconds MINIMUM ... before
my cell rings. And that is over and above the usual PDD to Bell Atlantic.

Why do I put up with it? Two reasons ... 1) no other choice and 2) most
time the Caller-ID is provided so I can call back the party that has just
given up on me because of the long PDD.

Just think of how many SS7 messages and voice trunk seizures were
required for the present PSTN in this case ..... every hop "set up" requires
signaling ... in this case between three service provider (trust) boundaries.

How can the RTT for going back to the home network (for the simple, no
special features case, only once) be worse than what I have now? And I
consider what I have now a useful service ... considering the alternative.

Every replacement technology challenges some of the "requirements"
upon which the present generation of technology is built upon. Optimizing
PDD for the PSTN-to-PSTN phone call was one of these requirements.
And while I still agree that minimizing PDD with some reasonableness of
the tradeoffs involved is still a laudable goal ... one should compare the
PDD tradeoffs of todays mobile/cellular world to indeed verify that what
we propose is -  in many cases - ALREADY BETTER than some usual
mobility cases (like my experience above).

As for the loss arguments .... I also agree with Jonathan. Moreover, we
have taken steps in IETF/sigtran to create transports that do not have the
head of line blocking and do have "fast retransmit" capability that our
more time sensitive signaling messages require (i.e., SCTP).

>
>
>> 
>> Morever, if we start from the assumption (for the above mentioned
>> reasons) that all SIP
>> signalling goes through at least one proxy, the advantages of having a
>> local proxy is
>> relevant for example in case of (SIP, hard) handovers (I' m always
>> referring to Henning' s approach to SIP mobility). The reduced RTT
>> results in a much smaller number of lost packets.
>
>How is that? Its a dangerous game to make strong assumptions on loss
>models like this.
>
>In any case, I think that optimizing an architecture based on
>assumptions about loss distributions and latency is somewhat
>short-sighted. Lousy network quality is going to kill you on the
>compressed voice traffic long before the signaling gets really blown
>away. Even from a human factors perspective, most people will tolerate
>long PDDs but not high voice packet loss rates. If the signaling loss is
>really bothering you, deploy good networks with diffserv giving great
>service to port 5060. Eventually these network performance problems will
>get solved. But, fundamental architectural decisions don't get changed.
>The ability to choose anyone as my applications provider, and not suffer
>horrible voice quality (since the audio always goes direct) is such a
>huge win for competition and service innovation. Saying no to that to
>save 50ms on signaling latency seems such a shame. This was my point
>about web. If I had to *rely* on my ISP replicating the content in order
>to fetch it, there never would have been a web. It is only because
>anyone, anywhere on the Internet could set up shop, and anyone, anywhere
>could access it, that we had the huge growth in this service. 
>
>-Jonathan R.
>
>-- 
>Jonathan D. Rosenberg                       72 Eagle Rock Ave.
>Chief Scientist                             First Floor
>dynamicsoft                                 East Hanover, NJ 07936
>jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
>http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
>http://www.dynamicsoft.com
>
>
>_______________________________________________
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>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip
>


Michael A. Ramalho, Ph.D.
Cisco Systems
1802 Rue de la Port
Wall Township, NJ 07719-3784
Office email: mramalho@cisco.com
Personal email: mar42@cornell.edu
Office: +1.732.449.5762
Cell: +1.732.809.0188
Pager: +1.800.365.4578


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From: Giuseppe Ricagni <giuseppe.ricagni@italtel.it>
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Henning,

Henning Schulzrinne wrote:

> There seem to be two cases:
>
> 1) The callee is local and the caller is roaming. In that case, there's
> no reason to have the caller go through its remote (home) proxy for
> sending SIP requests (except if that home proxy performs, for example,
> authentication services).
>

right !
This also applies to roaming mobile-to-fixed, local endpoint.

Authentication is the real open issue at the moment, but what about having
a full authentication involving the home domain at registration time (when
I switch on the mobile), and a local authentication with the local proxy
when the endpoint sends the Invite ? Would this be feasible ?


>
> 2) The callee is roaming (independent of where the caller is). The SIP
> call has no choice but to go the home proxy of the callee first, since
> that's the only reference point for the caller. It has no clue that the
> callee is standing right next to him/her.

As long as I set up a brand new call, you are absolutely right.

In my handover example, anyway, I was sending a new Invite with the same
call identifier of an existing call just to update the IP address where my
correspondent should send voice traffic, and I was trying to reduce the
handoff delay to a minimum. The costraint I put was that all SIP messages
were going through at least 1 proxy.

Now, if both terminals are roaming, and they are in the same domain
(typical case of two friends on vacation), since there already is an
established call between the two, the local proxy sholud be able to route
the Invite message towards the correspondent host without involving the
home network.

Correct ?

Thank you
Giuseppe

--
----------------------------------------------------------
Giuseppe RICAGNI
System Architecture and Product Planning
Broadband Networks
Italtel
via Reiss Romoli - C10
20019 Castelletto di Settimo Milanese (MILANO)
ITALY

mailto:giuseppe.ricagni@italtel.it
----------------------------------------------------------





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        Giuseppe Ricagni <giuseppe.ricagni@italtel.it>,
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Subject: SIP architecture for Roaming mobiles (was RE: [SIP] Minutes of SIP 
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Dear All,

I thought about generating a new thread as our discussion had completely
diverged from the purpose of the previous thread

Jörgen Björkner wrote:

>
> The home proxy always should be within the signaling path, even though
> you
> have roamed into another network. This is very important from a value
> added
> service perspective.

there are four kind of value added services:

   * a) those who require user profile information
   * b) those who don' t  require user profile information, but to some
     extent they are not supported by the visited domain
   * c) those who don' t  require user profile information and they are
     implemented by the visited domain.
   * d) null services: in other words, basic calls tha do not request
     any kind of value added service

Case a) and b) require some kind of interaction with the home network.

C) and d) don' t.

>

[...]

> The reason to have the home proxy within the signaling path is to be
> able to
> offer services that is invoked before the call is set up. One example
> is a
> speed dial dial service that translates one request URI to something
> else
> depending on a users preferences.

This is a type a) service. Yes, you need to execute it in the home
network, unless you download the user profile at registration time AND
the visited network has implemented that specific service.

With respect to the method used to classify service request within the
four cathegories, a possibility is that  the home domain, at
registration time, sends information to the visited domain on the
requirements to meet  to locally execute a given set of services for
that specific user. If the visited domain supports such requirements ==>
those services can be executed locally.

This mechanism can even  be extended to the full download of the user
profile.

>

[...]

> I have understood that some groups proposes a solution where the
> services is
> downloaded to the visited network. The solution with downloaded
> services has
> one severe problem, the fact that the execution environment must be
> homogenous between the operators.

The real problem I see here is a secuity one: PNOs will never allow 3rd
pty code to run on thei mission-critical machines.

> This is very limiting for a operator that
> want to distinguish himself and attract customers by providing
> creative
> services, which is very important on todays competitive market. A
> successful
> service might need to retrive information from an information system
> that
> the users service provider has access to. This means that althouh the
> code
> will be able to execute when it is downloaded to the visited network
> so will
> the information that is needed not be accessible.

> I don't think this would affect the call setup time in any significant
> way,
> and users is today used to longer call setup times when the roam into
> another operators network.

Please see my othe messages

Giuseppe

>
> /Jörgen
>
> -----Original Message-----
> From: James Kempf
> To: Michael Thomas; Giuseppe Ricagni
> Cc: jdrosen@dynamicsoft.com; sip@lists.bell-labs.com; James Kempf;
> Michael
> Thomas
> Sent: 2000-05-16 14:05
> Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
>
> >Giuseppe Ricagni writes:
> > >    * possible optimizations (e.g.: signalling for a basic call to
> an
> >Australian
> > >      fixed network number from an Italian mobile roaming to
> Australia could
> >be
> > >      processed locally, without any "tromboning")
> >
> >   If during call setup there needs to be a round
> >   trip to the home network, it pretty much negates
> >   any advantage of proxying the call locally.
> >
>
> When you take roaming into account, there is an advantage. Suppose the
>
> Italian now in Australia moves. If the home proxy is used, the network
>
> has to go back to the home to roam. If a local proxy is used, there is
>
> reduced latency. With a local proxy, you pay the round trip cost
> to the home network just once.
>
> In addition, some people have been talking about using SIP for other
> kinds of in-call signalling. If the mobile has to go all the way
> back to the home proxy, there is simply too much latency.
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--

--
----------------------------------------------------------
Giuseppe RICAGNI
System Architecture and Product Planning
Broadband Networks
Italtel
via Reiss Romoli - C10
20019 Castelletto di Settimo Milanese (MILANO)
ITALY

mailto:giuseppe.ricagni@italtel.it
----------------------------------------------------------


--------------66FDDE02B0E4D01E2733D4A2
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Dear All,
<p>I thought about generating a new thread as our discussion had completely
diverged from the purpose of the previous thread
<p>J&ouml;rgen Bj&ouml;rkner wrote:
<blockquote TYPE=CITE>&nbsp;
<br>The home proxy always should be within the signaling path, even though
you
<br>have roamed into another network. This is very important from a value
added
<br>service perspective.</blockquote>
there are four kind of value added services:
<ul>
<li>
a) those who require user profile information</li>

<li>
b) those who don' t&nbsp; require user profile information, but to some
extent they are not supported by the visited domain</li>

<li>
c) those who don' t&nbsp; require user profile information and they are
implemented by the visited domain.</li>

<li>
d) null services: in other words, basic calls tha do not request any kind
of value added service</li>
</ul>
Case a) and b) require some kind of interaction with the home network.
<p>C) and d) don' t.
<blockquote TYPE=CITE>&nbsp;</blockquote>
[...]
<blockquote TYPE=CITE>The reason to have the home proxy within the signaling
path is to be able to
<br>offer services that is invoked before the call is set up. One example
is a
<br>speed dial dial service that translates one request URI to something
else
<br>depending on a users preferences.</blockquote>
This is a type a) service. Yes, you need to execute it in the home network,
unless you download the user profile at registration time AND the visited
network has implemented that specific service.
<p>With respect to the method used to classify service request within the
four cathegories, a possibility is that&nbsp; the home domain, at registration
time, sends information to the visited domain on the requirements to meet&nbsp;
to locally execute a given set of services for that specific user. If the
visited domain supports such requirements ==> those services can be executed
locally.
<p>This mechanism can even&nbsp; be extended to the full download of the
user profile.
<blockquote TYPE=CITE>&nbsp;</blockquote>
[...]
<blockquote TYPE=CITE>I have understood that some groups proposes a solution
where the services is
<br>downloaded to the visited network. The solution with downloaded services
has
<br>one severe problem, the fact that the execution environment must be
<br>homogenous between the operators.</blockquote>
The real problem I see here is a secuity one: PNOs will never allow 3rd
pty code to run on thei mission-critical machines.
<blockquote TYPE=CITE>This is very limiting for a operator that
<br>want to distinguish himself and attract customers by providing creative
<br>services, which is very important on todays competitive market. A successful
<br>service might need to retrive information from an information system
that
<br>the users service provider has access to. This means that althouh the
code
<br>will be able to execute when it is downloaded to the visited network
so will
<br>the information that is needed not be accessible.</blockquote>

<blockquote TYPE=CITE>I don't think this would affect the call setup time
in any significant way,
<br>and users is today used to longer call setup times when the roam into
<br>another operators network.</blockquote>
Please see my othe messages
<p>Giuseppe
<blockquote TYPE=CITE>&nbsp;
<br>/J&ouml;rgen
<p>-----Original Message-----
<br>From: James Kempf
<br>To: Michael Thomas; Giuseppe Ricagni
<br>Cc: jdrosen@dynamicsoft.com; sip@lists.bell-labs.com; James Kempf;
Michael
<br>Thomas
<br>Sent: 2000-05-16 14:05
<br>Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
<p>>Giuseppe Ricagni writes:
<br>> >&nbsp;&nbsp;&nbsp; * possible optimizations (e.g.: signalling for
a basic call to an
<br>>Australian
<br>> >&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; fixed network number from an Italian
mobile roaming to
<br>Australia could
<br>>be
<br>> >&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; processed locally, without any "tromboning")
<br>>
<br>>&nbsp;&nbsp; If during call setup there needs to be a round
<br>>&nbsp;&nbsp; trip to the home network, it pretty much negates
<br>>&nbsp;&nbsp; any advantage of proxying the call locally.
<br>>
<p>When you take roaming into account, there is an advantage. Suppose the
<br>Italian now in Australia moves. If the home proxy is used, the network
<br>has to go back to the home to roam. If a local proxy is used, there
is
<br>reduced latency. With a local proxy, you pay the round trip cost
<br>to the home network just once.
<p>In addition, some people have been talking about using SIP for other
<br>kinds of in-call signalling. If the mobile has to go all the way
<br>back to the home proxy, there is simply too much latency.
<p>_______________________________________________
<br>SIP mailing list
<br>SIP@lists.bell-labs.com
<br><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a>
<p>_______________________________________________
<br>SIP mailing list
<br>SIP@lists.bell-labs.com
<br><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></blockquote>
--
<p>--
<br>----------------------------------------------------------
<br>Giuseppe RICAGNI
<br>System Architecture and Product Planning
<br>Broadband Networks
<br>Italtel
<br>via Reiss Romoli - C10
<br>20019 Castelletto di Settimo Milanese (MILANO)
<br>ITALY
<p><a href="mailto:giuseppe.ricagni@italtel.it">mailto:giuseppe.ricagni@italtel.it</a>
<br>----------------------------------------------------------
<br>&nbsp;</html>

--------------66FDDE02B0E4D01E2733D4A2--



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From sip-admin@lists.bell-labs.com  Wed May 17 11:56:35 2000
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Date: Wed, 17 May 2000 11:56:20 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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Just as a data point: Tony Eyers and I have looked at signaling
performance under redirect and proxy assumptions, for both ("optimized")
TCP and UDP. See the paper in the IPtel 2000 workshop at
http://www.fokus.gmd.de/events/iptel2000/pg.php3

This is based on recent real Internet measurements, assuming that SIP
will not use either reserved flows or BBE diff-serv classes. If one were
to use standard TCP with its long initial retransmit timer, things get
really bad (we didn't look at this since it's too bad to be
interesting...), but even with cross-country redirection, call setup
delays are rather modest. Note that we looked at percentiles, not
averages.


"Michael A. Ramalho" wrote:
> 

> FWIW - I would love only 5 seconds!
> 
> My home office phone (Bell Atlantic) is often forwarded to my AT&T Wireless
> phone. However, I'm usually in a Comcast serving area (i.e., roaming).
> 
> When I am at home when someone calls my forwarded home office
> phone ... it pings (to let me know a call has been forwarded) ... then I
> wait .... and wait some more ..... usually 6 to 8 seconds MINIMUM ... before
> my cell rings. And that is over and above the usual PDD to Bell Atlantic.
> 
> Why do I put up with it? Two reasons ... 1) no other choice and 2) most
> time the Caller-ID is provided so I can call back the party that has just
> given up on me because of the long PDD.
> 
> Just think of how many SS7 messages and voice trunk seizures were
> required for the present PSTN in this case ..... every hop "set up" requires
> signaling ... in this case between three service provider (trust) boundaries.
> 
> How can the RTT for going back to the home network (for the simple, no
> special features case, only once) be worse than what I have now? And I
> consider what I have now a useful service ... considering the alternative.
> 
> Every replacement technology challenges some of the "requirements"
> upon which the present generation of technology is built upon. Optimizing
> PDD for the PSTN-to-PSTN phone call was one of these requirements.
> And while I still agree that minimizing PDD with some reasonableness of
> the tradeoffs involved is still a laudable goal ... one should compare the
> PDD tradeoffs of todays mobile/cellular world to indeed verify that what
> we propose is -  in many cases - ALREADY BETTER than some usual
> mobility cases (like my experience above).
> 
> As for the loss arguments .... I also agree with Jonathan. Moreover, we
> have taken steps in IETF/sigtran to create transports that do not have the
> head of line blocking and do have "fast retransmit" capability that our
> more time sensitive signaling messages require (i.e., SCTP).
> 


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


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Giuseppe Ricagni wrote:
> 
> Henning,
> 
> Henning Schulzrinne wrote:
> 
> > There seem to be two cases:
> >
> > 1) The callee is local and the caller is roaming. In that case, there's
> > no reason to have the caller go through its remote (home) proxy for
> > sending SIP requests (except if that home proxy performs, for example,
> > authentication services).
> >
> 
> right !
> This also applies to roaming mobile-to-fixed, local endpoint.
> 
> Authentication is the real open issue at the moment, but what about having
> a full authentication involving the home domain at registration time (when
> I switch on the mobile), and a local authentication with the local proxy
> when the endpoint sends the Invite ? Would this be feasible ?

I should have been more precise: we need to distinguish roaming between
domains (from, say, columbia.edu to stanford.edu) and roaming between
proxies that share information/data (say, detroit.att.com and
nyc.att.com). In the latter case, presumably the Detroit server has
access to my NY "home" entry and can perform all normal outbound proxy
functions, including any outbound scripts, address books or other
personalized logic. The tricky part is how the UA decides what to do. We
have three local-proxy mechanisms:

- DHCP
- multicast
- local configuration (hard-wired)

The first two would presumably point to stanford.edu or detroit.att.com
as I roam away from Columbia, while the third is likely to be always
nyc.att.com.  With the DHCP option, the UA could guess as to whether to
use the home proxy or not, based on domain name. It would have to know,
somehow, whether the proxy is sharing information with home or not.


> In my handover example, anyway, I was sending a new Invite with the same
> call identifier of an existing call just to update the IP address where my
> correspondent should send voice traffic, and I was trying to reduce the
> handoff delay to a minimum. The costraint I put was that all SIP messages
> were going through at least 1 proxy.

See our SIP mobility paper and other more recent work that addresses
this, at a high level. We still need to work out the details as to how
this work, as it requires that the "away" proxy proxying the
registration modifies the Contact to point to it rather than the mobile,
for example.

> 
> Now, if both terminals are roaming, and they are in the same domain
> (typical case of two friends on vacation), since there already is an
> established call between the two, the local proxy sholud be able to route
> the Invite message towards the correspondent host without involving the
> home network.
> 
> Correct ?

Again, this depends on naming and whether a local outbound proxy allows
"foreign" registrations. If I visit Stanford and the Stanford proxy
allows me to register with my cs.columbia.edu address AND somehow
short-circuits INVITEs from a Stanford caller to me (using my
cs.columbia.edu address), this avoids the trip home, but has obviously
bad consequences if I'm relying on my cs.columbia.edu proxy to do call
filtering for me. Thus, I may not particularly appreciate that *unless*
I can get the Stanford proxy to inherit my call logic. Currently, I can
only do this by uploading the CPL script, say, which means my mobile has
to carry it around. One could presumably allow a level of indirection
where I register at Stanford with my a pointer to my canonical master
CPL script.

I doubt whether this complexity is worth the minor saving in call setup
delay. 

Thus, just about anything can be done, depending on naming, but each
choice has consequences on available call features.

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


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Date: Wed, 17 May 2000 16:58:07 +0200
From: Stephen Terrill <stephen.terrill@era.ericsson.se>
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
To: Giuseppe Ricagni <giuseppe.ricagni@italtel.it>
Cc: James.Kempf@Eng.Sun.COM, Michael Thomas <mat@cisco.com>,
        sip@lists.bell-labs.com, Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Hi Giuseppe,

One correction, in the deployed SS7 networks today, there is only a single 64kbs pipe available to the call (the signalling for one call is not split over several signalling links).  As such, the faster transport does reduce the latency - and I don´t believe that anyone is considering running the signalling on an improperly dimensioned/configured network.

Regards,

..//steve

Giuseppe Ricagni wrote:

> Stephen,
>
> Stephen Terrill wrote:
>
> > Hi James,
> >
> > I think that the emphansis on saving the trombing of the user signalling is much overweighted.  Firstly, the "singlling" transport in these networks is not limited to the 64kbs singnalling channels used in the PSTN today.
>
> Higher bandwidth doesn' t necessarily mean lower latency..... Moreover
> you don' t normally have *just one* 64Kb/s link. In this
> specific case, the tight delay requirements on SS7 networks, might well
> result in lower RTTs for signalling messages.
>
> > Secondly, even today, it is not uncommon for two operators in the same country (ie a european country) to interconnect to each other by going through the US due to commercial reasons (ie a call from one phone to another phone just 30 meeters away can cross the atlantic twice).  What I am saying as that this happens today, and that is both user signalling and payload, and customer usually aren´t aware of it.
> >
>
> not saying that signalling should *always* be processed locally: as you
> know this is just impossible, especially when user profile data is
> required to process the call.
>
> What we are claiming here is that local procesing should be employed as
> an optimization, in all those cases where it is actually possible,
> without any (pre-call) data exchange with the home domain.
>
> > As such, I do tend to agree with Jonathan and Mike that the advantages of taking the call/session signalling to the "home" proxy outweigh this small and dissappearing disadvantage.
>
> I do tend to disagree... Why not applying an optimization (only in those
> cases where it can be applied, obviously) if it's easy, it has basically
> no disadvantages, and it can be even much more helpful in other contexts
> (seen my other posting ?)
>
> Cheers
> Giuseppe
> P.S.
> BTW: you will remember that  “Optimised call routing” is a requirement
> for 3G networks (it does also refer to signalling)....
>
> >
> >
> > Regards,
> >
> > ..//steve
> >
> > James Kempf wrote:
> >
> > > >James Kempf writes:
> > > > > I don't think the analogy between SIP and HTTP is valid. I see it more
> > > > > as being between SIP and routing. Suppose I would have to go all the
> > > > > way back to my home network to get my packets routed.
> > > >
> > > >   This is a pretty dubious analogy. We're talking
> > > >   about proxy-routed signaling traffic and that
> > > >   alone. You're making it sound like the media
> > > >   would do dog-leg routing through the home
> > > >   network which is absurd.
> > > >
> > >
> > > No, that's not what I meant. Thhe point I was trying to make is that
> > > I don't think people will be very happy if it takes them
> > > substantially longer to make a telephone connection remotely than it
> > > does at home. It can take somewhat longer (as it does today in the
> > > circuit switched world) but not *substantially* longer. I don't have
> > > the numbers for intercontinental routing delays in my head but I can
> > > believe they must be more than routing delays to a local ISP. Whether
> > > those delays will, in fact, be substantial enough to merit having
> > > local proxies is a matter that can be solved by engineering analysis
> > > (and not political posturing, as this thread seems to have degenerated into).
> > >
> > > > > Even if you are willing to grant the analogy, HTTP requests nowadays often
> > > > > don't get routed to the server that somebody set up as their web site.
> > > > > If your ISP is concerned about service, they may have set up an Inktomi
> > > > > cache, and if the publisher is concerned, they may have contracted with
> > > > > Akami to do the same. So the world is not so simple as "let's make it
> > > > > free and easy for anybody to set up service" on the Web anymore.
> > > >
> > > >   You're assuming that every thing is going to
> > > >   be a gigantic commercial service. I disagree.
> > > >   It's still perfectly possible for me to run a
> > > >   web server at home, and in fact I do. We
> > > >   haven't engineered walls around that
> > > >   possibility. Requiring that I use a foreign
> > > >   proxy does, or at least thinks it does.
> > > >
> > >
> > > Your web server at home is leveraging off the fact that a company,
> > > probably AT&T back when they had a monopoly, strung a wire to your
> > > house and installed switching equipment at the local office. They made the infrastructue
> > > investment years ago that now allows you and your ISP to offer service
> > > really cheaply. You may be limited as to the bandwidth you get, due
> > > to the fact that your current local telephone provider is not willing
> > > to make the infrastructure investment to upgrade to fiber.
> > >
> > > The UK government just auctioned off UMTS licenses for something like $25B
> > > (as in 1 with 9 zeros behind it). Anyone who is not able to come up
> > > with that kind of cash will have a hard time offering wireless
> > > telephony service. It will take years to amortize that kind of investment.
> > > Of course, there is always the unlicensed spectrum and I'm sure there
> > > will be players like Metricom who figure out creative ways to utilize it.
> > >
> > > >   Also: you're talking about transparent caching,
> > > >   not proxying in the web case. I've not heard
> > > >   of any ISP that *require* that I go through
> > > >   their HTTP proxies.
> > > >
> > > > > You can
> > > > > set up your service, but unless you have a concerned ISP or the money
> > > > > to pay Akami, the latency in using your service may cause people to
> > > > > go elsewhere.
> > > >
> > > >   This analogy is fatally flawed. Cache's are
> > > >   great for the mountains of .gif's. There aren't
> > > >   similar considerations for SIP. Also: even
> > > >   if you're talking about co-lo servers
> > > >   distributed geographically, the site owner
> > > >   still "owns" the site. What you're talking
> > > >   about here is something run by their
> > > >   *competition* on your subscriber's behalf.
> > > >
> > > >   Yuck.
> > > >
> > >
> > > Cellular companies already have romaing agreements. There's nothing new about
> > > it.
> > >
> > > With regard to caches v.s. proxies, the only point I was trying to make was
> > > that customers want fast service. Whether that service is web page access
> > > or connecting for a telephone call. It therefore makes sense to engineer
> > > such that the service provider can provide that service, whether it requires
> > > transparent caches or local proxies.
> > >
> > > > > I know this isn't going to be a popular viewpoint, but I think we need
> > > > > to be realistic about the design. SIP already contains support for
> > > > > telling an incoming caller that a client has moved, I can't see the
> > > > > harm in using a remote proxy, given the performance concerns.
> > > >
> > > >   I think we have to demonstrate that there
> > > >   are legitimate performance concerns before we
> > > >   solve for them.
> > > >
> > >
> > > Agreed.
> > >
> > > >   Sanity check: we're talking about a
> > > >   hypothetical case where the SIP UA is
> > > >   in the handset, right? We're not talking
> > > >   about the case where you're terminating SIP
> > > >   at the next logical hop up (BTS, MSC) for
> > > >   legacy, right? I'm talking about the former.
> > > >   In the latter, its debatable which proxy
> > > >   should do routing.
> > > >
> > >
> > > I was talking about both. The legacy situtation will be important for
> > > some time, particularly in the North American case where the 2G and
> > > 3G spectrum allocations overlap. But I think the core network design
> > > should be there for smooth transition to the SIP based handset.
> > >
> > >                         jak
> > >
> > > _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > http://lists.bell-labs.com/mailman/listinfo/sip
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
>
> --
> ----------------------------------------------------------
> Giuseppe RICAGNI
> System Architecture and Product Planning
> Broadband Networks
> Italtel
> via Reiss Romoli - C10
> 20019 Castelletto di Settimo Milanese (MILANO)
> ITALY
>
> mailto:giuseppe.ricagni@italtel.it
> ----------------------------------------------------------
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip




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From sip-admin@lists.bell-labs.com  Wed May 17 12:38:20 2000
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I read in RFC2543bis, "SIP does not offer conference control services such as floor control or voting and does not prescribe how a conference is to be managed, but SIP can be used to introduce conference control protocols." This is a little confusing because in Abstract of RFC2543bis I read that these sessions(which are created, modified, and terminated using SIP), could also be Internet multimedia conferences.....

I skimmed through the document "Distributed Multipoint Conferences using SIP"(draft-mark-sip-dmcs-00). I've a situation where a central server does all the conferencing and I'd like SIP UACs to be able to estabilish a multimedia conference session on that server. e.g. for a conference call, they can specify hostname or IP address of the Conferencing Sever in the SDP and establish the session. 
Do we have an extension or something, which explains Conferencing on a central server? Any pointers, explanations, or directions would very appreciated.
Thanks,
Mohammad Vakil






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Dean Willis writes:
 > Actually, I think there might be an exception. Let's say we have a large network that mixes several DiffServe classes -- for simplicity, let's say "best effort" and "priority" clases. Assume that there is normally a relatively high ratio of best-effort to priority traffic in the network.
 > 
 > One might a heuristic for traffic engineering that works like this:
 > 
 > Assume a network monitoring system that can
 > provide both a rough estimate of the ration of
 > best-effort to priority traffic on the backbone
 > and on specific links. Feed this data into access
 > proxies.

   Doesn't this violate the law of "You are not
   alone"?

   Besides, even if it were just EF RTP traffic
   controlled by SIP proxies, how would they
   synchronize amongst themselves?

   Even with the implicit linkage model of
   signaling and admission authorization of
   DQoS, it's still the edge routers making the 
   admission control decision from a "can we
   support the bandwidth" standpoint into the
   diffserv cloud.

	       Mike


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Hi Stephen,

Stephen Terrill wrote:

> Hi Giuseppe,
>
> One correction, in the deployed SS7 networks today, there is only a single 64kbs pipe available to the call (the signalling for one call is not split over several signalling links).  As such, the faster transport does reduce the latency - and I don´t believe that anyone is considering running the signalling on an improperly dimensioned/configured network.

well, I just thought that by claiming that current SS7 signalling is conveyed on 64 Kbit/s links you were suggesting   some kind of conjestion *in the nodes* (STPs).

The average ISUP message length, just to make an example, does not normally exceed 40bytes.

A 64 Kbit/s link sends a 40 bytes message in 5 milliseconds: I really don' t think this is the bottleneck.....

Should you take it to 100 Mbit/s you wouln't gain definetly more than 5 msec !!!!

OTOH, the time spent in routers may differ from that spent in STPs, retransmission mechanisms may differ, and so on.

Giuseppe

>
> Regards,
>
> ..//steve
>
> Giuseppe Ricagni wrote:
>
> > Stephen,
> >
> > Stephen Terrill wrote:
> >
> > > Hi James,
> > >
> > > I think that the emphansis on saving the trombing of the user signalling is much overweighted.  Firstly, the "singlling" transport in these networks is not limited to the 64kbs singnalling channels used in the PSTN today.
> >
> > Higher bandwidth doesn' t necessarily mean lower latency..... Moreover
> > you don' t normally have *just one* 64Kb/s link. In this
> > specific case, the tight delay requirements on SS7 networks, might well
> > result in lower RTTs for signalling messages.
> >
> > > Secondly, even today, it is not uncommon for two operators in the same country (ie a european country) to interconnect to each other by going through the US due to commercial reasons (ie a call from one phone to another phone just 30 meeters away can cross the atlantic twice).  What I am saying as that this happens today, and that is both user signalling and payload, and customer usually aren´t aware of it.
> > >
> >
> > not saying that signalling should *always* be processed locally: as you
> > know this is just impossible, especially when user profile data is
> > required to process the call.
> >
> > What we are claiming here is that local procesing should be employed as
> > an optimization, in all those cases where it is actually possible,
> > without any (pre-call) data exchange with the home domain.
> >
> > > As such, I do tend to agree with Jonathan and Mike that the advantages of taking the call/session signalling to the "home" proxy outweigh this small and dissappearing disadvantage.
> >
> > I do tend to disagree... Why not applying an optimization (only in those
> > cases where it can be applied, obviously) if it's easy, it has basically
> > no disadvantages, and it can be even much more helpful in other contexts
> > (seen my other posting ?)
> >
> > Cheers
> > Giuseppe
> > P.S.
> > BTW: you will remember that  “Optimised call routing” is a requirement
> > for 3G networks (it does also refer to signalling)....
> >
> > >
> > >
> > > Regards,
> > >
> > > ..//steve
> > >
> > > James Kempf wrote:
> > >
> > > > >James Kempf writes:
> > > > > > I don't think the analogy between SIP and HTTP is valid. I see it more
> > > > > > as being between SIP and routing. Suppose I would have to go all the
> > > > > > way back to my home network to get my packets routed.
> > > > >
> > > > >   This is a pretty dubious analogy. We're talking
> > > > >   about proxy-routed signaling traffic and that
> > > > >   alone. You're making it sound like the media
> > > > >   would do dog-leg routing through the home
> > > > >   network which is absurd.
> > > > >
> > > >
> > > > No, that's not what I meant. Thhe point I was trying to make is that
> > > > I don't think people will be very happy if it takes them
> > > > substantially longer to make a telephone connection remotely than it
> > > > does at home. It can take somewhat longer (as it does today in the
> > > > circuit switched world) but not *substantially* longer. I don't have
> > > > the numbers for intercontinental routing delays in my head but I can
> > > > believe they must be more than routing delays to a local ISP. Whether
> > > > those delays will, in fact, be substantial enough to merit having
> > > > local proxies is a matter that can be solved by engineering analysis
> > > > (and not political posturing, as this thread seems to have degenerated into).
> > > >
> > > > > > Even if you are willing to grant the analogy, HTTP requests nowadays often
> > > > > > don't get routed to the server that somebody set up as their web site.
> > > > > > If your ISP is concerned about service, they may have set up an Inktomi
> > > > > > cache, and if the publisher is concerned, they may have contracted with
> > > > > > Akami to do the same. So the world is not so simple as "let's make it
> > > > > > free and easy for anybody to set up service" on the Web anymore.
> > > > >
> > > > >   You're assuming that every thing is going to
> > > > >   be a gigantic commercial service. I disagree.
> > > > >   It's still perfectly possible for me to run a
> > > > >   web server at home, and in fact I do. We
> > > > >   haven't engineered walls around that
> > > > >   possibility. Requiring that I use a foreign
> > > > >   proxy does, or at least thinks it does.
> > > > >
> > > >
> > > > Your web server at home is leveraging off the fact that a company,
> > > > probably AT&T back when they had a monopoly, strung a wire to your
> > > > house and installed switching equipment at the local office. They made the infrastructue
> > > > investment years ago that now allows you and your ISP to offer service
> > > > really cheaply. You may be limited as to the bandwidth you get, due
> > > > to the fact that your current local telephone provider is not willing
> > > > to make the infrastructure investment to upgrade to fiber.
> > > >
> > > > The UK government just auctioned off UMTS licenses for something like $25B
> > > > (as in 1 with 9 zeros behind it). Anyone who is not able to come up
> > > > with that kind of cash will have a hard time offering wireless
> > > > telephony service. It will take years to amortize that kind of investment.
> > > > Of course, there is always the unlicensed spectrum and I'm sure there
> > > > will be players like Metricom who figure out creative ways to utilize it.
> > > >
> > > > >   Also: you're talking about transparent caching,
> > > > >   not proxying in the web case. I've not heard
> > > > >   of any ISP that *require* that I go through
> > > > >   their HTTP proxies.
> > > > >
> > > > > > You can
> > > > > > set up your service, but unless you have a concerned ISP or the money
> > > > > > to pay Akami, the latency in using your service may cause people to
> > > > > > go elsewhere.
> > > > >
> > > > >   This analogy is fatally flawed. Cache's are
> > > > >   great for the mountains of .gif's. There aren't
> > > > >   similar considerations for SIP. Also: even
> > > > >   if you're talking about co-lo servers
> > > > >   distributed geographically, the site owner
> > > > >   still "owns" the site. What you're talking
> > > > >   about here is something run by their
> > > > >   *competition* on your subscriber's behalf.
> > > > >
> > > > >   Yuck.
> > > > >
> > > >
> > > > Cellular companies already have romaing agreements. There's nothing new about
> > > > it.
> > > >
> > > > With regard to caches v.s. proxies, the only point I was trying to make was
> > > > that customers want fast service. Whether that service is web page access
> > > > or connecting for a telephone call. It therefore makes sense to engineer
> > > > such that the service provider can provide that service, whether it requires
> > > > transparent caches or local proxies.
> > > >
> > > > > > I know this isn't going to be a popular viewpoint, but I think we need
> > > > > > to be realistic about the design. SIP already contains support for
> > > > > > telling an incoming caller that a client has moved, I can't see the
> > > > > > harm in using a remote proxy, given the performance concerns.
> > > > >
> > > > >   I think we have to demonstrate that there
> > > > >   are legitimate performance concerns before we
> > > > >   solve for them.
> > > > >
> > > >
> > > > Agreed.
> > > >
> > > > >   Sanity check: we're talking about a
> > > > >   hypothetical case where the SIP UA is
> > > > >   in the handset, right? We're not talking
> > > > >   about the case where you're terminating SIP
> > > > >   at the next logical hop up (BTS, MSC) for
> > > > >   legacy, right? I'm talking about the former.
> > > > >   In the latter, its debatable which proxy
> > > > >   should do routing.
> > > > >
> > > >
> > > > I was talking about both. The legacy situtation will be important for
> > > > some time, particularly in the North American case where the 2G and
> > > > 3G spectrum allocations overlap. But I think the core network design
> > > > should be there for smooth transition to the SIP based handset.
> > > >
> > > >                         jak
> > > >
> > > > _______________________________________________
> > > > SIP mailing list
> > > > SIP@lists.bell-labs.com
> > > > http://lists.bell-labs.com/mailman/listinfo/sip
> > >
> > > _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > http://lists.bell-labs.com/mailman/listinfo/sip
> >
> > --
> > ----------------------------------------------------------
> > Giuseppe RICAGNI
> > System Architecture and Product Planning
> > Broadband Networks
> > Italtel
> > via Reiss Romoli - C10
> > 20019 Castelletto di Settimo Milanese (MILANO)
> > ITALY
> >
> > mailto:giuseppe.ricagni@italtel.it
> > ----------------------------------------------------------
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip

--
----------------------------------------------------------
Giuseppe RICAGNI
System Architecture and Product Planning
Broadband Networks
Italtel
via Reiss Romoli - C10
20019 Castelletto di Settimo Milanese (MILANO)
ITALY

mailto:giuseppe.ricagni@italtel.it
----------------------------------------------------------





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From sip-admin@lists.bell-labs.com  Wed May 17 13:44:19 2000
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To: <sip@lists.research.bell-labs.com>
Cc: <Mohammad.Vakil@outreachtech.com>
Subject: RE: [SIP] Multimedia Conferencing using SIP
Date: Wed, 17 May 2000 10:28:00 -0700
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Mohammad,

The now expired call control draft (draft-ietf-mmusic-sip-cc-01) describes
a SIP bridge which would accomplish the goals you wish describing how a
UAC could establish a connection to a centralized conference server
without any SIP extensions.  See section 4.3 of the draft.  Here's a link:

http://www.softarmor.com/sipwg/drafts/draft-ietf-mmusic-sip-cc-01.txt

It's also important to note the last 2 paragraphs of this section describe
a mechanism for one participant in the bridged conference to INVITE
another to the conference.  This mechanism is still up in the air and is
work in progress.

-Jeff

--
Jeff Mark                               Intel Dialogic Division
Software Engineer                       201 Mentor Dr, Suite 100
mailto:Jeff.Mark@dialogic.com           Santa Barbara, CA 93111
http://www.dialogic.com                 (805) 879-3640
                                        (805) 967-1395 fax


> -----Original Message-----
> From: Vakil, Mohammad [mailto:MVakil@outreachtech.com]
> Sent: Wednesday, May 17, 2000 9:22 AM
> To: sip@lists.research.bell-labs.com
> Cc: 'Mohammad.Vakil@outreachtech.com'
> Subject: [SIP] Multimedia Conferencing using SIP
>
>
>
> I read in RFC2543bis, "SIP does not offer conference control
> services such as floor control or voting and does not
> prescribe how a conference is to be managed, but SIP can be
> used to introduce conference control protocols." This is a
> little confusing because in Abstract of RFC2543bis I read
> that these sessions(which are created, modified, and
> terminated using SIP), could also be Internet multimedia
> conferences.....
>
> I skimmed through the document "Distributed Multipoint
> Conferences using SIP"(draft-mark-sip-dmcs-00). I've a
> situation where a central server does all the conferencing
> and I'd like SIP UACs to be able to estabilish a multimedia
> conference session on that server. e.g. for a conference
> call, they can specify hostname or IP address of the
> Conferencing Sever in the SDP and establish the session.
> Do we have an extension or something, which explains
> Conferencing on a central server? Any pointers, explanations,
> or directions would very appreciated.
> Thanks,
> Mohammad Vakil
>
>



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James Kempf writes:
 > >   I know they do, and my understanding is that
 > >   feature transparency is one thing that suffers
 > >   big time because of their current model. 
 > >   Keeping my features back where I get my 
 > >   features eliminates that feature synch
 > >   problem at its root. That is impossible in
 > >   the circuit switched world, but almost trivial
 > >   in the SIP world.
 > >
 > 
 > If features can be easily forwarded between proxies (with the proper
 > authentication) then it's even easier to get your features whereever
 > you want them.

   That is a monumentally large "if". I know
   of no system which has solved that
   satisfactorily on any reasonable scale.
   Throwing in trust boundaries, and different
   competing vendors installations is just gravy.
   I'd claim that this is an intractible
   problem. Some people might believe that 
   this is solvable, but I remain extremely
   skeptical. IMO, the best way to deal with
   intractible problems is to steer well clear
   of them.

	Mike


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From sip-admin@lists.bell-labs.com  Wed May 17 15:34:25 2000
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From sip-admin@lists.bell-labs.com  Wed May 17 17:25:13 2000
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Date: Wed, 17 May 2000 17:24:29 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Gethin Liddell wrote:
> 
> hi
> 
> I have a problem with the In-Reply-To header, 2543 bis section 6.25 p. 48:
> 
> "The In-Reply-To request header field enumerates the call-IDs that this call references or returns.
> This allows automatic call distribution systems to route return calls to the originator of the first call and allows
> callees to filter calls, so that only calls that return calls they have originated will be accepted. This field is not a
> substitute for request authentication.
> 
> In-Reply-To =  In-Reply-To   :  1# callid"
> 
> If i can refer you to Anders email titled:
> 
> Re: [SIP] Another grammar query - Call-ID
> 
> then callid has been re-defined as
> "Basically, the field should be treated as an opaque,
> uninterpreted identifier, e.g. TEXT-UTF8-TRIM in the grammar."
> 
> The problem is that if the In-Reply-To header is:
> In-Reply-To: 70710@saturn.bell-tel.com, 17320@saturn.bell-tel.com
> 
> then this will resolve as a single call-id not two separate ones.
> 
> would it not be possible to say make call-id a quoted-string???
> This may make little sense for the Call-ID header but it solves the problem here.
> what would we then compare on though?  could escaped values get in here which would
> stop the efficient byte by byte equality checking that TEXT-UTF8-TRIM provided?

This raises a broader issue. I see no reason why call-IDs should need to
be escaped in some uses, i.e., allowing any UTF-8 string seems overkill.
Call-IDs are just random identifiers not human-generated text, so
there's no reason to allow commas, etc. Allowing "strange" characters
just makes it more difficult for log files and anywhere else they might
appear.

Indeed, earlier we had agreed to use 'token' | 'token' @ 'token' or
something like that as the appropriate BNF entity.


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


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From: "Dean Willis" <dean.willis@softarmor.com>
To: "IETF SIP" <sip@lists.bell-labs.com>, "Michael Thomas" <mat@cisco.com>
Subject: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task force in Adelaide)
Date: Wed, 17 May 2000 17:16:02 -0500
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Comments inline

Mike ponders:
> Dean Willis writes:
>  > Actually, I think there might be an exception. Let's say we 
> have a large network that mixes several DiffServe classes -- for 
> simplicity, let's say "best effort" and "priority" clases. Assume 
> that there is normally a relatively high ratio of best-effort to 
> priority traffic in the network.
>  > 
>  > One might a heuristic for traffic engineering that works like this:
>  > 
>  > Assume a network monitoring system that can
>  > provide both a rough estimate of the ration of
>  > best-effort to priority traffic on the backbone
>  > and on specific links. Feed this data into access
>  > proxies.
> 
>    Doesn't this violate the law of "You are not
>    alone"?

We're not making assurances -- we're making guesses about the state of the network. The question: Does the core network look like it will support another EF flow right now? The law of large numbers provides an averaging effect here. Remember, this is a big, densely-connected core.

> 
>    Besides, even if it were just EF RTP traffic
>    controlled by SIP proxies, how would they
>    synchronize amongst themselves?

They don't need to. One call makes little difference to the core network. We just define the admissions threshold well into the "safe" area. Statistical distribution takes care of the rest.

> 
>    Even with the implicit linkage model of
>    signaling and admission authorization of
>    DQoS, it's still the edge routers making the 
>    admission control decision from a "can we
>    support the bandwidth" standpoint into the
>    diffserv cloud.

Different edge -- it's the edge of carrier network vs. the edge of customer network. And the feedback is through network management to SIP, and it could be completely decoupled from a DQOS.

Consider it in an aggregation model, like a donut. The chewy outside parts of the donut are controlled by IntServ. Traffic leaps across the DiffServ donut-hole only if the heuristic process thinks (IE, is reasonably sure) that there is adequate bandwidth.

--
Dean

ÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÒ ùšŠX§‚X¬µ"þX¬¶ÏÛzYÿ•¦ìýÊ&†ÛiÿÿåŠËlý·¥—ùZnÏÜ¢oæj)fjåŠËbú?


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Subject: [SIP] another SIP question
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--------------AD3DAA1ECC8F6D032328A73C
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Hello,

    My question concerns the following scenario: When user A calls user
B,  the proxy server would forward the request to user B, in case user B
is not available, the proxy server might opt to put the call to user B's
mailbox. My question is that how does the mailbox
know whom the incoming call is intended for, is there an existing  field
to hold the information?

   regards,



-- Jimmy zhang (jz@ipunity.com)

   Software Engineer



--------------AD3DAA1ECC8F6D032328A73C
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Hello,
<p>&nbsp;&nbsp;&nbsp; My question concerns the following scenario: When
user A calls user B,&nbsp; the proxy server would forward the request to
user B, in case user B is not available, the proxy server might opt to
put the call to user B's mailbox. My question is that how does the mailbox
<br>know whom the incoming call is intended for, is there an existing&nbsp;
field to hold the information?
<p>&nbsp;&nbsp; regards,
<br>&nbsp;
<br>&nbsp;
<pre>-- Jimmy zhang (jz@ipunity.com)

&nbsp;&nbsp; Software Engineer</pre>
&nbsp;</html>

--------------AD3DAA1ECC8F6D032328A73C--



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From sip-admin@lists.bell-labs.com  Thu May 18 01:23:04 2000
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Date: Thu, 18 May 2000 01:30:42 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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        Sip Mail List <sip@lists.bell-labs.com>
Subject: Re: [SIP] Problem with definition of In-Reply-To grammar
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Henning Schulzrinne wrote:
> 
> > The problem is that if the In-Reply-To header is:
> > In-Reply-To: 70710@saturn.bell-tel.com, 17320@saturn.bell-tel.com
> >
> > then this will resolve as a single call-id not two separate ones.
> >
> > would it not be possible to say make call-id a quoted-string???
> > This may make little sense for the Call-ID header but it solves the problem here.
> > what would we then compare on though?  could escaped values get in here which would
> > stop the efficient byte by byte equality checking that TEXT-UTF8-TRIM provided?
> 
> This raises a broader issue. I see no reason why call-IDs should need to
> be escaped in some uses, i.e., allowing any UTF-8 string seems overkill.
> Call-IDs are just random identifiers not human-generated text, so
> there's no reason to allow commas, etc. Allowing "strange" characters
> just makes it more difficult for log files and anywhere else they might
> appear.
> 
> Indeed, earlier we had agreed to use 'token' | 'token' @ 'token' or
> something like that as the appropriate BNF entity.

I agree completely. There is no reason to make life more complicated
here.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Thu May 18 01:41:25 2000
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Giuseppe Ricagni wrote:
> 

> I think we are talking about two different things.
> I was just sayng that, according to Henning's model, to put it very simple, when you change
> IP address you shuld send an Invite to your correspondent host containing your new IP
> address (plus re-registering with your proxy, to allow for signalling to be sent to the new
> address).
> 
> If we assume (by hipotesys) that all SIP messages must go through at least 1 proxy, and if
> we have an active conversation between two Italian mobiles located in Australia, the so
> called "handoff delay" is one RTT between Australia and Italy, which can be something like
> 300 msec or (very likely) more.
> Even with a 300msec figure, you would en up losing some 15 voice samples, which is something
> definetly undesirable.
> 
> OTOH, with a local proxy, you wouldn' t lose more than 1 or, at worst, two samples.

This presumes that there aren't any other proxies which decide to
record-route. When a call is made, it will generally pass through many
proxies, each providing differing levels of service. Some of these will
want to record-route. I don't want to prevent them from doing so.
Optimizing a system for the special case where (1) both caller and
called party have home domains that are far, (2) they are both roaming
in visiting domains that are close, (3) there are no other proxies on
the path which record-route, seems like you are optimizing for the wrong
case.

Your assumption is also that handover is done through a re-INVITE to the
new address once its known. This need not be the case. In a scenario
with a smoother handover, I can imagine a brief period where the roaming
mobile has both addresses - one in the old domain, one in the new. When
it obtains the new one, it does a re-INVITE to *add* a new media stream
targeted to the new address. This means it will get both stream of
packets during this transition period. Once these new packets start
arriving, it can do a re-INVITE again, turning off the stream targeted
to the old address. It can then release the old address.

I admit ignorance in not knowing if such a smooth handover at the IP
level is possible. But, if so, it completely decouples the signaling RTT
from any packet loss during handovers.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
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Date: Thu, 18 May 2000 01:41:11 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Cc: sip@lists.research.bell-labs.com,
        "'Mohammad.Vakil@outreachtech.com'" <Mohammad.Vakil@outreachtech.com>
Subject: Re: [SIP] Multimedia Conferencing using SIP
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"Vakil, Mohammad" wrote:
> 
> I read in RFC2543bis, "SIP does not offer conference control services such as floor control or voting and does not prescribe how a conference is to be managed, but SIP can be used to introduce conference control protocols." This is a little confusing because in Abstract of RFC2543bis I read that these sessions(which are created, modified, and terminated using SIP), could also be Internet multimedia conferences.....


Don't confuse multimedia conferences (i.e., voice plus video muliparty
things) and conference control services. The latter include, as the bis
draft states, floor control (who can talk now), and voting. SIP doesn't
do conference control, but it can certainly initiate multimedia
multiparty sessions.

> 
> I skimmed through the document "Distributed Multipoint Conferences using SIP"(draft-mark-sip-dmcs-00). I've a situation where a central server does all the conferencing and I'd like SIP UACs to be able to estabilish a multimedia conference session on that server. e.g. for a conference call, they can specify hostname or IP address of the Conferencing Sever in the SDP and establish the session.
> Do we have an extension or something, which explains Conferencing on a central server? Any pointers, explanations, or directions would very appreciated.

No extensions needed for central conference servers. It works just like
it does in the PSTN - you dial a number which corresponds to a port on a
bridge. In the case of SIP, you would INVITE to a URL corresponding to a
port on the conference server, like sip:conference33287372@mcu.com. The
result appears as a point to point SIP call as far as  your UA is
concerned, but the server is mixing media and sending the mixed media to
your UA. You will learn the identity of the other participants through
RTCP CSRC.

How do you know the right URL to send an INVITE to? Well, how do you
know what number to dial for a conference in PSTN? You're told out of
band. URLs can be included in emails, web pages, whatever.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Thu May 18 05:23:56 2000
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Date: Thu, 18 May 2000 11:22:17 +0200
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
From: Romeo Zwart <Romeo.Zwart@icoe.att.com>
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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Hi Henning,

At 11:56 AM 17-05-00 -0400, you wrote:
>Just as a data point: Tony Eyers and I have looked at signaling
>performance under redirect and proxy assumptions, for both ("optimized")
>TCP and UDP. See the paper in the IPtel 2000 workshop at
>http://www.fokus.gmd.de/events/iptel2000/pg.php3

These measurements did not include the PRACK model. Do you
know if anyone is working on a followup that addresses the reliable
provisional response model as well?

>This is based on recent real Internet measurements, assuming that SIP
>will not use either reserved flows or BBE diff-serv classes. If one were
>to use standard TCP with its long initial retransmit timer, things get
>really bad (we didn't look at this since it's too bad to be
>interesting...), but even with cross-country redirection, call setup
>delays are rather modest. Note that we looked at percentiles, not
>averages.

This is a very interesting simulation. Has a more detailed analysis
of these results been published? Some of the results seem rather
counter-intuitive.

For example, if you compare graphs 12 (simple call NY-Chi) and 14
(redirect, NY-West Coast), the minimum setup delay roughly triples.
Considering the combined effect of the added redirect and the larger
distance, this is a trend you might expect, albeit a rather large increase.
Comparing the same graphs on the 95 percentile scores, the result
seems to be that variation increases a lot, which again seems to
make sense. However the graphs also suggests that the trend is
downward, rather than upward, i.e. from roughly 600 ms in the
simple call case, down to 200 ms on many of the observed days
for the redirected Coast-to-Coast case.


Romeo



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Cc: sip@lists.bell-labs.com, Tony Eyers <tony@elec.uow.edu.au>
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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Romeo Zwart wrote:
> 
> Hi Henning,
> 
> At 11:56 AM 17-05-00 -0400, you wrote:
> >Just as a data point: Tony Eyers and I have looked at signaling
> >performance under redirect and proxy assumptions, for both ("optimized")
> >TCP and UDP. See the paper in the IPtel 2000 workshop at
> >http://www.fokus.gmd.de/events/iptel2000/pg.php3
> 
> These measurements did not include the PRACK model. Do you
> know if anyone is working on a followup that addresses the reliable
> provisional response model as well?

Not that I know of, but PRACK would seem to be a fairly limited-use
model. You'd have to address also the whole QoS setup mechanism, I
suppose...


> 
> This is a very interesting simulation. Has a more detailed analysis
> of these results been published? Some of the results seem rather
> counter-intuitive.

Not sure if you looked at the paper or the slides, but the paper is
about as complete as we have at this point.

> 
> For example, if you compare graphs 12 (simple call NY-Chi) and 14
> (redirect, NY-West Coast), the minimum setup delay roughly triples.
> Considering the combined effect of the added redirect and the larger
> distance, this is a trend you might expect, albeit a rather large increase.
> Comparing the same graphs on the 95 percentile scores, the result
> seems to be that variation increases a lot, which again seems to
> make sense. However the graphs also suggests that the trend is
> downward, rather than upward, i.e. from roughly 600 ms in the
> simple call case, down to 200 ms on many of the observed days
> for the redirected Coast-to-Coast case.

Tony, want to comment?
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
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From: tony@elec.uow.edu.au
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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Romeo, Henning,
              As you say, the results that you mention appear
counter-intuitive. A detailed discussion appears in section
VII of the paper. However, the reason for the apparent discrepancy
(i.e. the New York-Chicago path having a higher 95th delay percentile
than the longer New York-West Coast one) is the consistently higher
error rates seen on the New York-Chicago path. These errors cause
SIP timeouts, and hence push the 95th percentile over 500 msec.

As you mention, the graphs show minimum delays which
reflect the respective propagation delays expected over the
two paths. The 95th percentile results show two
features: the delay arising from the 500 msec SIP timeouts,
and the delay arising from queueing variations
(e.g. the results around 200-300 msec on the New York-West
Coast graph).

In general, the 95th percentile of the call setup delay
is greatly influenced by the prevailing error rate. We also
do show that increasing the number of hops in the call
setup path can greatly increase the 95th delay percentile
(e.g. figures 9 and 10 from the paper). 

Hope this helps.

Regards,

Tony
At 07:27 AM 5/18/00 -0400, Henning Schulzrinne wrote:
>Romeo Zwart wrote:
>> 
>> Hi Henning,
>> 
>> At 11:56 AM 17-05-00 -0400, you wrote:
>> >Just as a data point: Tony Eyers and I have looked at signaling
>> >performance under redirect and proxy assumptions, for both ("optimized")
>> >TCP and UDP. See the paper in the IPtel 2000 workshop at
>> >http://www.fokus.gmd.de/events/iptel2000/pg.php3
>> 
>> These measurements did not include the PRACK model. Do you
>> know if anyone is working on a followup that addresses the reliable
>> provisional response model as well?
>
>Not that I know of, but PRACK would seem to be a fairly limited-use
>model. You'd have to address also the whole QoS setup mechanism, I
>suppose...
>
>
>> 
>> This is a very interesting simulation. Has a more detailed analysis
>> of these results been published? Some of the results seem rather
>> counter-intuitive.
>
>Not sure if you looked at the paper or the slides, but the paper is
>about as complete as we have at this point.
>
>> 
>> For example, if you compare graphs 12 (simple call NY-Chi) and 14
>> (redirect, NY-West Coast), the minimum setup delay roughly triples.
>> Considering the combined effect of the added redirect and the larger
>> distance, this is a trend you might expect, albeit a rather large increase.
>> Comparing the same graphs on the 95 percentile scores, the result
>> seems to be that variation increases a lot, which again seems to
>> make sense. However the graphs also suggests that the trend is
>> downward, rather than upward, i.e. from roughly 600 ms in the
>> simple call case, down to 200 ms on many of the observed days
>> for the redirected Coast-to-Coast case.
>
>Tony, want to comment?
>-- 
>Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
>



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Jonathan Rosenberg wrote:

> Giuseppe Ricagni wrote:
> >
>
> > I think we are talking about two different things.
> > I was just sayng that, according to Henning's model, to put it very simple, when you change
> > IP address you shuld send an Invite to your correspondent host containing your new IP
> > address (plus re-registering with your proxy, to allow for signalling to be sent to the new
> > address).
> >
> > If we assume (by hipotesys) that all SIP messages must go through at least 1 proxy, and if
> > we have an active conversation between two Italian mobiles located in Australia, the so
> > called "handoff delay" is one RTT between Australia and Italy, which can be something like
> > 300 msec or (very likely) more.
> > Even with a 300msec figure, you would en up losing some 15 voice samples, which is something
> > definetly undesirable.
> >
> > OTOH, with a local proxy, you wouldn' t lose more than 1 or, at worst, two samples.
>
> This presumes that there aren't any other proxies which decide to
> record-route. When a call is made, it will generally pass through many
> proxies, each providing differing levels of service. Some of these will
> want to record-route.I don't want to prevent them from doing so.

Basically, the only problem here is that the callee is roaming.

As Henning reminded us, this requires a database (let' s call it Home Subscriber Server: HSS, a
kind of GSM HLR) lookup by the "calling proxy" in the callee home domain, to have information
about the address of the proxy the callee is registered at.

This lookup can be performed directly by the caller proxy, or via a callee home domain "home
proxy".

This  "home proxy" is the only proxy you would like to write in the record-route and I would like
not to.

IMHO, since  from a technical point of view you could query the HSS directly from the calling
proxy, you are to involving the "home proxy" only for "political reasons", to some extent to
"filter" queries to the HSS.

There is therefore no point in having  such proxy in the record-route.

Also remember we are talking here about basic calls, requesting no additional services, or
services that   are user profile independent. More complex scenarios can be achieved, as I
suggested in a previous posting:


there are four kind of value added services:

     a) those who require user profile information
     b) those who don' t  require user profile information, but to some extent they are not
     supported by the visited domain
     c) those who don' t  require user profile information and they are implemented by the
     visited domain.
     d) null services: in other words, basic calls that do not request any kind of value added
     service

Case a) and b) require some kind of interaction with the home network.

C) and d) don' t.

[....]

type a) services. need to execute it in the home network, unless you download
the user profile at registration time AND the visited network has implemented that specific
service.

With respect to the method used to classify service request within the four cathegories, a
possibility is that  the home domain, at registration time, sends information to the visited
domain
on the requirements to meet  to locally execute a given set of services for that specific user. If

the visited domain supports such requirements ==> those services can be executed locally.

This mechanism can even  be extended to the full download of the user profile.


> Optimizing a system for the special case where (1) both caller and
> called party have home domains that are far, (2) they are both roaming
> in visiting domains that are close, (3) there are no other proxies on
> the path which record-route, seems like you are optimizing for the wrong
> case.

First of all: we are not "optimizing" a system, here: we are adding a feature, that can be useful
to "optimize" a given set of cases.

Let's make a simpler  (and very common) example:
an Italian mobile roaming to Australia, that wants to call a (local !) Taxy: is this the wrong
case ? (In that case I'm wrong myself, because I did it in Adelaide!!!  :-))

It is a basic call, no supplementary services required, it can definetly handled locally. The
calle is a PSTN number, therefore no HSS lookup is neded (well, even if the calee was an
australian mobile phone, it would have been a local -Australian- lookup of the Australian HSS).
Why should I send my Invite to Italy ?!?!?


> Your assumption is also that handover is done through a re-INVITE to the
> new address once its known.

well, something like that. (the Invite is sent FROM the mobile host -the one changing address-
towards its correspondent)

> This need not be the case. In a scenario
> with a smoother handover, I can imagine a brief period where the roaming
> mobile has both addresses - one in the old domain, one in the new. When
> it obtains the new one, it does a re-INVITE to *add* a new media stream
> targeted to the new address. This means it will get both stream of
> packets during this transition period. Once these new packets start
> arriving, it can do a re-INVITE again, turning off the stream targeted
> to the old address. It can then release the old address.

Sorry, but I think you missed the target.

The problem is only related to the in-flight packets.
If I' m in the position of keeping the two address active for a while, I will get the in-flight
packets on the old
address anyway !!!! I don' t need any duplicte invite !!!

The problem comes when I lose the old address and there are some already in-flight packets
addressed to it.
The number of in-flight packets is proportional to the time needed to make  the other endpoint
aware of the new address.
No duplicate Invites or whatever can help.

Giuseppe

>
> I admit ignorance in not knowing if such a smooth handover at the IP
> level is possible. But, if so, it completely decouples the signaling RTT
> from any packet loss during handovers.
>
> -Jonathan R.
>
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com

--
----------------------------------------------------------
Giuseppe RICAGNI
System Architecture and Product Planning
Broadband Networks
Italtel
via Reiss Romoli - C10
20019 Castelletto di Settimo Milanese (MILANO)
ITALY

mailto:giuseppe.ricagni@italtel.it
----------------------------------------------------------


--------------FC6D76027E7D29F668A93CDE
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Jonathan Rosenberg wrote:
<blockquote TYPE=CITE>Giuseppe Ricagni wrote:
<br>>
<p>> I think we are talking about two different things.
<br>> I was just sayng that, according to Henning's model, to put it very
simple, when you change
<br>> IP address you shuld send an Invite to your correspondent host containing
your new IP
<br>> address (plus re-registering with your proxy, to allow for signalling
to be sent to the new
<br>> address).
<br>>
<br>> If we assume (by hipotesys) that all SIP messages must go through
at least 1 proxy, and if
<br>> we have an active conversation between two Italian mobiles located
in Australia, the so
<br>> called "handoff delay" is one RTT between Australia and Italy, which
can be something like
<br>> 300 msec or (very likely) more.
<br>> Even with a 300msec figure, you would en up losing some 15 voice
samples, which is something
<br>> definetly undesirable.
<br>>
<br>> OTOH, with a local proxy, you wouldn' t lose more than 1 or, at worst,
two samples.
<p>This presumes that there aren't any other proxies which decide to
<br>record-route. When a call is made, it will generally pass through many
<br>proxies, each providing differing levels of service. Some of these
will
<br>want to record-route.I don't want to prevent them from doing so.</blockquote>

<p><br>Basically, the only problem here is that the callee is roaming.
<p>As Henning reminded us, this requires a database (let' s call it Home
Subscriber Server: HSS, a kind of GSM HLR) lookup by the "calling proxy"
in the callee home domain, to have information about the address of the
proxy the callee is registered at.
<p>This lookup can be performed directly by the caller proxy, or via a
callee home domain "home proxy".
<p>This&nbsp; "home proxy" is the only proxy you would like to write in
the record-route and I would like not to.
<p>IMHO, since&nbsp; from a technical point of view you could query the
HSS directly from the calling proxy, you are to involving the "home proxy"
only for "political reasons", to some extent to "filter" queries to the
HSS.
<p>There is therefore no point in having&nbsp; such proxy in the record-route.
<p>Also remember we are talking here about basic calls, requesting no additional
services, or services that&nbsp;&nbsp; are user profile independent. More
complex scenarios can be achieved, as I suggested in a previous posting:
<br>&nbsp;
<p><font color="#3333FF">there are four kind of value added services:</font>
<p><font color="#3333FF">&nbsp;&nbsp;&nbsp;&nbsp; a) those who require
user profile information</font>
<br><font color="#3333FF">&nbsp;&nbsp;&nbsp;&nbsp; b) those who don' t&nbsp;
require user profile information, but to some extent they are not</font>
<br><font color="#3333FF">&nbsp;&nbsp;&nbsp;&nbsp; supported by the visited
domain</font>
<br><font color="#3333FF">&nbsp;&nbsp;&nbsp;&nbsp; c) those who don' t&nbsp;
require user profile information and they are implemented by the</font>
<br><font color="#3333FF">&nbsp;&nbsp;&nbsp;&nbsp; visited domain.</font>
<br><font color="#3333FF">&nbsp;&nbsp;&nbsp;&nbsp; d) null services: in
other words, basic calls that do not request any kind of value added</font>
<br><font color="#3333FF">&nbsp;&nbsp;&nbsp;&nbsp; service</font>
<p><font color="#3333FF">Case a) and b) require some kind of interaction
with the home network.</font>
<p><font color="#3333FF">C) and d) don' t.</font>
<p><font color="#3333FF">[....]</font>
<p><font color="#3333FF">type a) services. need to execute it in the home
network, unless you download</font>
<br><font color="#3333FF">the user profile at registration time AND the
visited network has implemented that specific</font>
<br><font color="#3333FF">service.</font>
<p><font color="#3333FF">With respect to the method used to classify service
request within the four cathegories, a</font>
<br><font color="#3333FF">possibility is that&nbsp; the home domain, at
registration time, sends information to the visited domain</font>
<br><font color="#3333FF">on the requirements to meet&nbsp; to locally
execute a given set of services for that specific user. If</font>
<br><font color="#3333FF">the visited domain supports such requirements
==> those services can be executed locally.</font>
<p><font color="#3333FF">This mechanism can even&nbsp; be extended to the
full download of the user profile.</font>
<br>&nbsp;
<blockquote TYPE=CITE>Optimizing a system for the special case where (1)
both caller and
<br>called party have home domains that are far, (2) they are both roaming
<br>in visiting domains that are close, (3) there are no other proxies
on
<br>the path which record-route, seems like you are optimizing for the
wrong
<br>case.</blockquote>
First of all: we are not "optimizing" a system, here: <b><u>we are adding
a feature</u></b>, that can be useful to "optimize" a given set of cases.
<p>Let's make a simpler&nbsp; (and very common) example:
<br>an Italian mobile roaming to Australia, that wants to call a (local
!) Taxy: is this the wrong case ? (In that case I'm wrong myself, because
I did it in Adelaide!!!&nbsp; :-))
<p>It is a basic call, no supplementary services required, it can definetly
handled locally. The calle is a PSTN number, therefore no HSS lookup is
neded (well, even if the calee was an australian mobile phone, it would
have been a local -Australian- lookup of the Australian HSS).
<br>Why should I send my Invite to Italy ?!?!?
<br>&nbsp;
<blockquote TYPE=CITE>Your assumption is also that handover is done through
a re-INVITE to the
<br>new address once its known.</blockquote>
well, something like that. (the Invite is sent FROM the mobile host -the
one changing address- towards its correspondent)
<blockquote TYPE=CITE>This need not be the case. In a scenario
<br>with a smoother handover, I can imagine a brief period where the roaming
<br>mobile has both addresses - one in the old domain, one in the new.
When
<br>it obtains the new one, it does a re-INVITE to *add* a new media stream
<br>targeted to the new address. This means it will get both stream of
<br>packets during this transition period. Once these new packets start
<br>arriving, it can do a re-INVITE again, turning off the stream targeted
<br>to the old address. It can then release the old address.</blockquote>
Sorry, but I think you missed the target.
<p>The problem is only related to the in-flight packets.
<br>If I' m in the position of keeping the two address active for a while,
I will get the in-flight packets on the old
<br>address anyway !!!! I don' t need any duplicte invite !!!
<p>The problem comes when I lose the old address and there are some already
in-flight packets addressed to it.
<br>The number of in-flight packets is proportional to the time needed
to make&nbsp; the other endpoint aware of the new address.
<br>No duplicate Invites or whatever can help.
<p>Giuseppe
<blockquote TYPE=CITE>&nbsp;
<br>I admit ignorance in not knowing if such a smooth handover at the IP
<br>level is possible. But, if so, it completely decouples the signaling
RTT
<br>from any packet loss during handovers.
<p>-Jonathan R.
<p>--
<br>Jonathan D. Rosenberg&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
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>jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
FAX:&nbsp;&nbsp; (732) 741-4778
<br><a href="http://www.cs.columbia.edu/~jdrosen">http://www.cs.columbia.edu/~jdrosen</a>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
PHONE: (732) 741-7244
<br><a href="http://www.dynamicsoft.com">http://www.dynamicsoft.com</a></blockquote>
--
<br>----------------------------------------------------------
<br>Giuseppe RICAGNI
<br>System Architecture and Product Planning
<br>Broadband Networks
<br>Italtel
<br>via Reiss Romoli - C10
<br>20019 Castelletto di Settimo Milanese (MILANO)
<br>ITALY
<p><a href="mailto:giuseppe.ricagni@italtel.it">mailto:giuseppe.ricagni@italtel.it</a>
<br>----------------------------------------------------------
<br>&nbsp;</html>

--------------FC6D76027E7D29F668A93CDE--



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Date: Thu, 18 May 2000 15:11:29 +0200
To: <tony@elec.uow.edu.au>
From: Romeo Zwart <Romeo.Zwart@icoe.att.com>
Subject: SIP Signalling delays (was Re: [SIP] Minutes of SIP Security
  task force in Adelaide)
Cc: Henning Schulzrinne <schulzrinne@cs.columbia.edu>, sip@lists.bell-labs.com
In-Reply-To: <3.0.6.32.20000518220519.009db100@elec.uow.edu.au>
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 <14620.41878.763300.49818@thomasm-u1.cisco.com>
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Henning, Tony,

Thanks for your explanations on the graphs. I admit having only
seen the presentation, not the paper. The proceedings are rather
difficult to obtain from the fokus website - I've restarted several
download attempts today, without much success. Could you
forward me a copy of the paper directly?

At 10:05 PM 18-05-00 +1000, tony@elec.uow.edu.au wrote:
[... stuff deleted ...]
>As you mention, the graphs show minimum delays which
>reflect the respective propagation delays expected over the
>two paths. The 95th percentile results show two
>features: the delay arising from the 500 msec SIP timeouts,
>and the delay arising from queueing variations
>(e.g. the results around 200-300 msec on the New York-West
>Coast graph).

OK, so the 500 ms timeout explains the 'quatum leap' effect
seen in the graphs - instead of a smooth distribution of setup
times. Which may suggest that packet loss contributes
significantly, even more than (queuing and propagation)
delay itself.

>In general, the 95th percentile of the call setup delay
>is greatly influenced by the prevailing error rate. We also
>do show that increasing the number of hops in the call
>setup path can greatly increase the 95th delay percentile
>(e.g. figures 9 and 10 from the paper).

If there is a correlation with the number of (router)hops, this
also suggests packet loss as a contributor, although queuing
delay can't be ignored, ofcourse.

I am very interested to have a look at your paper.

Regards,

Romeo

>Regards,
>
>Tony



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Subject: Re: SIP Signalling delays (was Re: [SIP] Minutes of SIP Securitytask 
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I've added the paper reference to
http://www.cs.columbia.edu/~hgs/sip/papers.html and
http://www.cs.columbia.edu/~hgs/research/imm/

Thanks for your interest.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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jimmy Zhang wrote:
> 
> Hello,
> 
>     My question concerns the following scenario: When user A calls
> user B,  the proxy server would forward the request to user B, in case
> user B is not available, the proxy server might opt to put the call to
> user B's mailbox. My question is that how does the mailbox
> know whom the incoming call is intended for, is there an existing
> field to hold the information?

The To field of the request always holds the original recipient the call
was destined for. It remains unchanged as the request is forwarded
through the network.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From: Kevin Summers <Kevin.Summers@ttimail.com>
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In draft-ietf-sip-isup-mime-01.txt, it is not explicitly stated where
encapsulation of an ISUP message should begin. It seems logical to begin
with the message type (as the example in the draft shows), but nowhere does
the draft prescribe exactly where to start.

I don't think that this decision should to be left to the implementors --
perhaps we should strengthen the requirements of how the message is
encapsulated. 
Any thoughts?

Kevin Summers


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From sip-admin@lists.bell-labs.com  Thu May 18 12:41:06 2000
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I am in the process of updating the "SIP Extensions for Media Authorization"
draft (
http://search.ietf.org/internet-drafts/draft-dcsgroup-sip-call-auth-01.txt).
Since there were no comments during IETF meeting and in the mailing list, I am
only planning to make some editorial cleaning and correct section 10 compliance.

If you have any questions and/or suggestions please let me know.

Burcak Beser

Burcak_Beser@3Com.com
3Com Carrier Systems Group




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        Michael Thomas <mat@cisco.com>, James Kempf <James.Kempf@Eng.Sun.COM>,
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Giuseppe Ricagni writes:
 > Basically, the only problem here is that the callee is roaming.
 > 
 > As Henning reminded us, this requires a database (let' s call it Home Subscriber Server: HSS, a
 > kind of GSM HLR) lookup by the "calling proxy" in the callee home domain, to have information
 > about the address of the proxy the callee is registered at.
 > 
 > This lookup can be performed directly by the caller proxy, or via a callee home domain "home
 > proxy".

   Sorry to be flippant, but gag me with a spoon.

   If you remote the proxy, this presupposes that:

   1) We have a standardized database transport protocol
   2) [far worse] we have a standardized database scheme

   versus:

   going directly to my home proxy for which I do not
   need to expose a database transport protocol or
   a scheme at all.

   This is a no-brainer to me: having to standardize
   a database scheme will stiffle innovation and 
   propogate the feature non-transparency brain damage
   of the current system. This is unacceptible.

   Let's take step back here though: the claimed
   reason for wanting a remote proxy is so that 
   you don't incur a transport layer dog leg clear
   back to my home proxy half way around the
   world. However, I'll claim that that problem
   can be solved far more adequately by putting
   proxies geographically closer to the roamed
   user. I doubt that it is unreasonable to assume
   that AT&T, etc, can place a home proxy for my
   service physically on Australia just like
   anybody else.
   
   The only challenge is how my mobile unit gets
   directed toward a home proxy that is
   geographically close to me (thus helping 
   round trip latencies). I'll claim that that
   is a useful problem to solve overall because
   web services, etc, have the identical problem.
   
   Maybe it's already solved for all I know. Even
   if it is not, I'd far rather work on that than
   have to come up with some grandiose, all
   encompassing database scheme. It is an
   intractable problem, and has many clear
   disadvantages for all parties.

		    Mike

   


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As a point of reference, I do hope that (some) services will be portable
across providers, if not for reasons of mobility, then for the ability
to take them with you when you change providers. CPL has that aim. It
would not be hard, as I indicated earlier, to have CPL scripts "roam"
with the user, without all agreeing on the universal LDAP schema or
database interface. Whether service providers will offer this feature is
probably less a technical than a business issue. I don't think it's a
good idea to rely on this being available, but service portability is a
worthwhile goal independent of this application. If a set of providers
agrees to support CPL, say, and then say "we switch your calls faster",
why not?
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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--------------23DF57763576BC8A139DFD1F
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Sorry Michael,

I think we are diverging from the original issue.

   * When the callee is roaming, either you use a standardized database transport protocol or you
     delegate the callee home domain proxy to perform the query, but you have a RTT towards the callee
     home domain. Your solution of having synchronized copies of such database all over the world is
     another interesting enhancement, maybe expensive (pretty sure off-the-shelf solutions are already
     out there)

        o the point I was making was that for all subsequent mid-call messages, just like the INVITEs
          you have to send for handovers, you should shortcut and avoid the round trip to the home
          net.


   * When the caller is roaming, under certain circumstances (related to the type of value added
     services required & the set of services supported by the visited network &  the possibility to
     transfer a user profile--see my previous messages) it is possible to avoid sending signalling
     back to the caller home domain. This applies to calls towards the fixed network (which may be
     routed by TRIP) and for calls to local mobiles. For roaming mobiles you fall back to the prvious
     case: still you don' t need to send signalling to the caller home domain, but you have a RTT to
     the callee home domain.

Hope this can summarize the discussion

Giuseppe

Michael Thomas wrote:

> Giuseppe Ricagni writes:
>  > Basically, the only problem here is that the callee is roaming.
>  >
>  > As Henning reminded us, this requires a database (let' s call it Home Subscriber Server: HSS, a
>  > kind of GSM HLR) lookup by the "calling proxy" in the callee home domain, to have information
>  > about the address of the proxy the callee is registered at.
>  >
>  > This lookup can be performed directly by the caller proxy, or via a callee home domain "home
>  > proxy".
>
>    Sorry to be flippant, but gag me with a spoon.
>
>    If you remote the proxy, this presupposes that:
>
>    1) We have a standardized database transport protocol
>    2) [far worse] we have a standardized database scheme
>
>    versus:
>
>    going directly to my home proxy for which I do not
>    need to expose a database transport protocol or
>    a scheme at all.
>
>    This is a no-brainer to me: having to standardize
>    a database scheme will stiffle innovation and
>    propogate the feature non-transparency brain damage
>    of the current system. This is unacceptible.
>
>    Let's take step back here though: the claimed
>    reason for wanting a remote proxy is so that
>    you don't incur a transport layer dog leg clear
>    back to my home proxy half way around the
>    world. However, I'll claim that that problem
>    can be solved far more adequately by putting
>    proxies geographically closer to the roamed
>    user. I doubt that it is unreasonable to assume
>    that AT&T, etc, can place a home proxy for my
>    service physically on Australia just like
>    anybody else.
>
>    The only challenge is how my mobile unit gets
>    directed toward a home proxy that is
>    geographically close to me (thus helping
>    round trip latencies). I'll claim that that
>    is a useful problem to solve overall because
>    web services, etc, have the identical problem.
>
>    Maybe it's already solved for all I know. Even
>    if it is not, I'd far rather work on that than
>    have to come up with some grandiose, all
>    encompassing database scheme. It is an
>    intractable problem, and has many clear
>    disadvantages for all parties.
>
>                     Mike
>
>

--
----------------------------------------------------------
Giuseppe RICAGNI
System Architecture and Product Planning
Broadband Networks
Italtel
via Reiss Romoli - C10
20019 Castelletto di Settimo Milanese (MILANO)
ITALY

mailto:giuseppe.ricagni@italtel.it
----------------------------------------------------------


--------------23DF57763576BC8A139DFD1F
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Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Sorry Michael,
<p>I think we are diverging from the original issue.
<ul>
<li>
When the <b><u>callee is roaming</u></b>, either you use a standardized
database transport protocol or you delegate the callee home domain proxy
to perform the query, but <b><u>you have a RTT</u></b> towards the callee
home domain. Your solution of having synchronized copies of such database
all over the world is another interesting enhancement, maybe expensive
(pretty sure off-the-shelf solutions are already out there)</li>
</ul>

<ul>
<ul>
<li>
the point I was making was that <b><u>for all subsequent mid-call messages</u></b>,
just like the INVITEs you have to send for handovers, <b><u>you should
shortcut and avoid the round trip</u></b> to the home net.</li>
</ul>

<br>&nbsp;
<li>
When the <b><u>caller is roaming</u></b>, under certain circumstances (related
to the type of value added services required &amp; the set of services
supported by the visited network &amp;&nbsp; the possibility to transfer
a user profile--see my previous messages) <b><u>it is possible to avoid
sending signalling back to the caller home</u></b> domain. This applies
to calls towards the fixed network (which may be routed by TRIP) and for
calls to local mobiles. For roaming mobiles you fall back to the prvious
case: still you don' t need to send signalling to the caller home domain,
but you have a RTT to the callee home domain.</li>
</ul>
Hope this can summarize the discussion
<p>Giuseppe
<p>Michael Thomas wrote:
<blockquote TYPE=CITE>Giuseppe Ricagni writes:
<br>&nbsp;> Basically, the only problem here is that the callee is roaming.
<br>&nbsp;>
<br>&nbsp;> As Henning reminded us, this requires a database (let' s call
it Home Subscriber Server: HSS, a
<br>&nbsp;> kind of GSM HLR) lookup by the "calling proxy" in the callee
home domain, to have information
<br>&nbsp;> about the address of the proxy the callee is registered at.
<br>&nbsp;>
<br>&nbsp;> This lookup can be performed directly by the caller proxy,
or via a callee home domain "home
<br>&nbsp;> proxy".
<p>&nbsp;&nbsp; Sorry to be flippant, but gag me with a spoon.
<p>&nbsp;&nbsp; If you remote the proxy, this presupposes that:
<p>&nbsp;&nbsp; 1) We have a standardized database transport protocol
<br>&nbsp;&nbsp; 2) [far worse] we have a standardized database scheme
<p>&nbsp;&nbsp; versus:
<p>&nbsp;&nbsp; going directly to my home proxy for which I do not
<br>&nbsp;&nbsp; need to expose a database transport protocol or
<br>&nbsp;&nbsp; a scheme at all.
<p>&nbsp;&nbsp; This is a no-brainer to me: having to standardize
<br>&nbsp;&nbsp; a database scheme will stiffle innovation and
<br>&nbsp;&nbsp; propogate the feature non-transparency brain damage
<br>&nbsp;&nbsp; of the current system. This is unacceptible.
<p>&nbsp;&nbsp; Let's take step back here though: the claimed
<br>&nbsp;&nbsp; reason for wanting a remote proxy is so that
<br>&nbsp;&nbsp; you don't incur a transport layer dog leg clear
<br>&nbsp;&nbsp; back to my home proxy half way around the
<br>&nbsp;&nbsp; world. However, I'll claim that that problem
<br>&nbsp;&nbsp; can be solved far more adequately by putting
<br>&nbsp;&nbsp; proxies geographically closer to the roamed
<br>&nbsp;&nbsp; user. I doubt that it is unreasonable to assume
<br>&nbsp;&nbsp; that AT&amp;T, etc, can place a home proxy for my
<br>&nbsp;&nbsp; service physically on Australia just like
<br>&nbsp;&nbsp; anybody else.
<p>&nbsp;&nbsp; The only challenge is how my mobile unit gets
<br>&nbsp;&nbsp; directed toward a home proxy that is
<br>&nbsp;&nbsp; geographically close to me (thus helping
<br>&nbsp;&nbsp; round trip latencies). I'll claim that that
<br>&nbsp;&nbsp; is a useful problem to solve overall because
<br>&nbsp;&nbsp; web services, etc, have the identical problem.
<p>&nbsp;&nbsp; Maybe it's already solved for all I know. Even
<br>&nbsp;&nbsp; if it is not, I'd far rather work on that than
<br>&nbsp;&nbsp; have to come up with some grandiose, all
<br>&nbsp;&nbsp; encompassing database scheme. It is an
<br>&nbsp;&nbsp; intractable problem, and has many clear
<br>&nbsp;&nbsp; disadvantages for all parties.
<p>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Mike
<br>&nbsp;
<br>&nbsp;</blockquote>
--
<br>----------------------------------------------------------
<br>Giuseppe RICAGNI
<br>System Architecture and Product Planning
<br>Broadband Networks
<br>Italtel
<br>via Reiss Romoli - C10
<br>20019 Castelletto di Settimo Milanese (MILANO)
<br>ITALY
<p><a href="mailto:giuseppe.ricagni@italtel.it">mailto:giuseppe.ricagni@italtel.it</a>
<br>----------------------------------------------------------
<br>&nbsp;</html>

--------------23DF57763576BC8A139DFD1F--



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Jonathan/Henning:

In RFC2543bis, section 2, 2nd last paragraph on page 17, you say:

   "The user parameter value "np-queried" indicates that the user part
    contains a telephone number AND that the number reflects the result
    of a query to the local number portability database".

However, in Figure 3 on page 18, the BNF for user-param is:

   user-param = "user="("phone" | "ip")

Should it be modified to:

   user-param = "user="(("phone" | "np-queried") | "ip")

A look at the archives indicate that Jonathan and Marc Petit-Huguenin
had a thread on this around May 10, 2000 with Jonathan stating that 
there was no consensus on if this was the right way to do this.  Any
updates?

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
IN Architecture and Internet Software Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184


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From sip-admin@lists.bell-labs.com  Thu May 18 15:25:25 2000
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To: Francois-Xavier Guitton <Francois-Xavier.Guitton@ms.alcatel.fr>
Cc: SIP <sip@lists.bell-labs.com>
Subject: Re: [SIP] Use of warning header in an INVITE message
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Francois-Xavier Guitton wrote:
> 
> Hello,
> 
> RFC 2543 specifies warning header as optional in the INVITE method.
> But the chapter describing Warning (6.41 Warning p 72) only details
> the usage of that header in case of a response. So my question is:
> what is the purpose of the warning header in an INVITE message ?
> 

This simply means it is optinal in a Warning header returned in response
to any request. Cf., for example, the description of the Contact header.

> 
> 

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


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To: Giuseppe Ricagni <giuseppe.ricagni@italtel.it>
Cc: Michael Thomas <mat@cisco.com>, sip@lists.bell-labs.com
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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Giuseppe Ricagni writes:
 >    * When the callee is roaming, either you use a standardized database transport protocol or you
 >      delegate the callee home domain proxy to perform the query, but you have a RTT towards the callee
 >      home domain. 

   Which may effectively be zero.

> Your solution of having synchronized copies of such database all over the world is
 >      another interesting enhancement, maybe expensive (pretty sure off-the-shelf solutions are already
 >      out there)

   It doesn't sound much more interesting than a
   localize LDAP cache on the client proxy. There
   are other ways to do this as well. And it doesn't
   imply synchronized; all that is necessary are caches
   with TTL's.

 >         o the point I was making was that for all subsequent mid-call messages, just like the INVITEs
 >           you have to send for handovers, you should shortcut and avoid the round trip to the home
 >           net.

   I disagree. Most traffic is between the two UA's.
   And it absolutely does not follow that proxies 
   need to have any part in handoffs. In fact, I
   do not believe that proxies have any part in
   handoffs when the UA is in the mobile unit. 

 >    * When the caller is roaming, under certain circumstances (related to the type of value added
 >      services required & the set of services supported by the visited network &  the possibility to
 >      transfer a user profile--see my previous messages)

   I don't know what services the roaming network
   wants to provide -- other than trying to shoehorn
   themselves into the call. Even if, there is nothing
   to prevent me from obtaining those service even if
   it's routed through my home proxy first.

	       Mike


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From sip-admin@lists.bell-labs.com  Thu May 18 15:34:14 2000
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Subject: Re: [SIP] URI's in SIP messages
References: <200005081932.OAA06333@b04a45.exu.ericsson.se> <39173AAA.E5B25BC7@americasm01.nt.com> <3919E388.A317ECAC@dynamicsoft.com> <391AB09F.917BD671@americasm01.nt.com> <391AF6C0.6C159A84@dynamicsoft.com>
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Jonathan Rosenberg wrote:
> 
>
> Right; I'm basically saying I believe this grammar to be wrong. URI
> should be absoluteURI.

Changed to:

Request-Line & = & Method SP Request-URI SP SIP-Version CRLF\\ 
Request-URI  & = & SIP-URI $|$ absoluteURI

where absoluteURI has been defined in RFC 2396.

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


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From sip-admin@lists.bell-labs.com  Thu May 18 16:00:08 2000
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Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
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Sorry Michael,

I really think we are looping: can we take it off-line ?

Giuseppe

Michael Thomas wrote:

> Giuseppe Ricagni writes:
>  >    * When the callee is roaming, either you use a standardized database transport protocol or you
>  >      delegate the callee home domain proxy to perform the query, but you have a RTT towards the callee
>  >      home domain.
>
>    Which may effectively be zero.
>
> > Your solution of having synchronized copies of such database all over the world is
>  >      another interesting enhancement, maybe expensive (pretty sure off-the-shelf solutions are already
>  >      out there)
>
>    It doesn't sound much more interesting than a
>    localize LDAP cache on the client proxy. There
>    are other ways to do this as well. And it doesn't
>    imply synchronized; all that is necessary are caches
>    with TTL's.
>
>  >         o the point I was making was that for all subsequent mid-call messages, just like the INVITEs
>  >           you have to send for handovers, you should shortcut and avoid the round trip to the home
>  >           net.
>
>    I disagree. Most traffic is between the two UA's.
>    And it absolutely does not follow that proxies
>    need to have any part in handoffs. In fact, I
>    do not believe that proxies have any part in
>    handoffs when the UA is in the mobile unit.
>
>  >    * When the caller is roaming, under certain circumstances (related to the type of value added
>  >      services required & the set of services supported by the visited network &  the possibility to
>  >      transfer a user profile--see my previous messages)
>
>    I don't know what services the roaming network
>    wants to provide -- other than trying to shoehorn
>    themselves into the call. Even if, there is nothing
>    to prevent me from obtaining those service even if
>    it's routed through my home proxy first.
>
>                Mike

--
----------------------------------------------------------
Giuseppe RICAGNI
System Architecture and Product Planning
Broadband Networks
Italtel
via Reiss Romoli - C10
20019 Castelletto di Settimo Milanese (MILANO)
ITALY

mailto:giuseppe.ricagni@italtel.it
----------------------------------------------------------




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Henning,

I assume this also applies to "addr-spec" in [6.15]. Also SIP-URI
currently called SIP-URL [2]; doesn't matter as long as it's consistant.

Rick

Henning Schulzrinne wrote:

> Jonathan Rosenberg wrote:
> >
> >
> > Right; I'm basically saying I believe this grammar to be wrong. URI
> > should be absoluteURI.
>
> Changed to:
>
> Request-Line & = & Method SP Request-URI SP SIP-Version CRLF\\
> Request-URI  & = & SIP-URI $|$ absoluteURI
>
> where absoluteURI has been defined in RFC 2396.
>
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

--------------E6ECBFE69F1E93845CBD7A3A
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<tt>Henning,</tt><tt></tt>
<p><tt>I assume this also applies to "addr-spec" in [6.15]. Also SIP-URI
currently called SIP-URL [2]; doesn't matter as long as it's consistant.</tt><tt></tt>
<p><tt>Rick</tt><tt></tt>
<p><tt>Henning Schulzrinne wrote:</tt>
<blockquote TYPE=CITE><tt>Jonathan Rosenberg wrote:</tt>
<br><tt>></tt>
<br><tt>></tt>
<br><tt>> Right; I'm basically saying I believe this grammar to be wrong.
URI</tt>
<br><tt>> should be absoluteURI.</tt><tt></tt>
<p><tt>Changed to:</tt><tt></tt>
<p><tt>Request-Line &amp; = &amp; Method SP Request-URI SP SIP-Version
CRLF\\</tt>
<br><tt>Request-URI&nbsp; &amp; = &amp; SIP-URI $|$ absoluteURI</tt><tt></tt>
<p><tt>where absoluteURI has been defined in RFC 2396.</tt><tt></tt>
<p><tt>--</tt>
<br><tt>Henning Schulzrinne&nbsp;&nbsp; <a href="http://www.cs.columbia.edu/~hgs">http://www.cs.columbia.edu/~hgs</a></tt></blockquote>
<tt></tt></html>

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Henning Schulzrinne writes:
 > There seem to be two cases:
 > 
 > 1) The callee is local and the caller is roaming. In that case, there's
 > no reason to have the caller go through its remote (home) proxy for
 > sending SIP requests (except if that home proxy performs, for example,
 > authentication services).

   That's a pretty big exception. I think there's
   a lot more reasons, like for example there is no
   standard way of the roamed proxy to know what my
   subscriber information is. Nor should there be any
   expectation that a roamed proxy would know what to
   do with that information except for a the barest
   least common denominator.

 > 2) The callee is roaming (independent of where the caller is). The SIP
 > call has no choice but to go the home proxy of the callee first, since
 > that's the only reference point for the caller. It has no clue that the
 > callee is standing right next to him/her.

   This no different than the non-roamed case.

	   Mike


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Cc: "IETF SIP" <sip@lists.bell-labs.com>, "Michael Thomas" <mat@cisco.com>
Subject: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task force in Adelaide)
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Dean Willis writes:
 > Consider it in an aggregation model, like a donut. The chewy outside parts of the donut are controlled by IntServ. Traffic leaps across the DiffServ donut-hole only if the heuristic process thinks (IE, is reasonably sure) that there is adequate bandwidth.

   OK, but I still don't see what that has to do
   with proxies. That sounds like something an
   edge router would do before it admits diffserv
   traffic into the diffserv core. Proxies are even
   more clueless because they never see the actual
   traffic patterns.

		Mike


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5.1.3, considering the case where the IWF does not contain a registrar or a
gatekeeper, says 

"If the destination address does not indicate the signaling protocol, a SIP
proxy server has to forward all incoming requests to a local IWF, just in
case the destination happens to be reachable via H.323."

Is this strictly true?  It would seem to me that a SIP proxy would first try
to find the called party using its normal SIP mechanisms, which would work
as usual for most cases (SIP-to-SIP, non-interworked calls).  Only if the
SIP proxy could not find the called party would it forward the call to an
H.323 IWF, hoping to find the destination in the H.323 cloud before
returning a 404 Not Found response.

4, requirement #2, says

"the user may dial any address without actually knowing whether it belongs
to the H.323 or the SIP cloud"

Is 'dial' referring to a SIP request to a tel URL?  Or just a request with a
more general destination (perhaps an alias) than a sip URL?

Tim Schroeder
SBC Technology Resources


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To: Kevin Summers <Kevin.Summers@ttimail.com>, sip@lists.bell-labs.com
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From: "Lyndon Ong" <long@nortelnetworks.com>
Subject: Re: [SIP] Payload Boundary of ISUP Object?
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Kevin,

Actually it mentions this in the Example, we could simply move this up to
Section 3.1 instead.

Cheers,

L. Ong

At 10:26 AM 5/18/2000 -0500, Kevin Summers wrote:
>In draft-ietf-sip-isup-mime-01.txt, it is not explicitly stated where
>encapsulation of an ISUP message should begin. It seems logical to begin
>with the message type (as the example in the draft shows), but nowhere does
>the draft prescribe exactly where to start.
>
>I don't think that this decision should to be left to the implementors --
>perhaps we should strengthen the requirements of how the message is
>encapsulated. 
>Any thoughts?
>
>Kevin Summers
>
>
>message scanned for virus content.
>
>telecom technologies, inc.
>http://www.ttiweb.com
>
>
>
>
>_______________________________________________
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>


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Still only in PostScript; see http://www.cs.columbia.edu/sip.
-- 
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From sip-admin@lists.bell-labs.com  Fri May 19 01:03:43 2000
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Hi,

According to the description of Content-Dispotion (C-D) header, its role is
to tell the UA whether the message (part) has to be
necessarily idenfied or not by the UA.

So, if I had a MIME header with 2 parts, with my first part being an ISUP
encoded payload and my second part being a message which I really dont care
if the UA understands or not,  I do:

--- begin ----
INVITE sip:01628434456@dcs1.nortelnetworks.com SIP/2.0
<etc ... etc>
Content-Type: multipart/mixed; boundary=unique-boundary-1

--unique-boundary-1
Content-type:application/isup
Content-Transfer-Encoding: binary

.. <binary payload > ...

--unique-boundary-1
Content-type:application/moohaa
Content-Diposition:inline;handling=optional

I dont really care if you understand me or not

--unique-boundary-1--

--- end ---

Is the above correct ?

I could find any reference to what the disposition type token is all abt,
and when I should specify which.
Also, what is the extension-param in disposition type for ?

If all its use is to tell the UA to mandatorily/optionally understand a
message part, then isnt

Content-Disposition - "Content-Disposition" ":" dispositon-type
disposition-type = "optional" | "required"
enough ?
Regds
Arjun

--
Arjun Roychowdhury @ Hughes Software Systems




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Hi All,

This is under the assumption that as a user moves during a conversation, the point of presence to the IP also moves.  In the popular radio access systems that I am aware of, this doesn´t happen, the "mico-mobility" (as it is sometimes refered to) is taken care of by the packer radio access network.

Regards,

..//Steve

Ashutosh Dutta wrote:

> As far as whether to talk to the home proxy or local proxy, for a wireless
> roaming user which is  changing domains frequently during mid-session-call and
> is on the move, it may be better to talk to the local proxy rather than going to
> the home proxy all the way during every hand-off (for latency reason I guess),
> of course there needs to be a security association between the local-proxy and
> home proxy or via public-proxy. This is one of the things we were trying to
> debate during SIP-2000 conference in Paris last week to see the trade-off
> between either talking to home-proxy or local domain proxy during frequent
> hand-off, then there is also a possibility of interaction between local SIP
> server and AAA server for a roaming user's profile verification during domain
> handoff.
>
> Thanks
> Ashutosh
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip



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As I said before, some services to set up a call might depend on information
that only is available in the callers home network, and for security or
policy reasons might be impossible to access from other networks. In this
case MUST the call setup signaling go through the home network.

I belive that this could be solved by adding a route header field containing
the address of the home server in the Invite message. If you follow the spec
must the first proxy in the visited network route the Invite to the home
proxy. This method allows for optimizing call setup when there is no service
that requires routing through the home network, and a method to indicate
when so should be the case.

/Jörgen

	  
>*	When the caller is roaming, under certain circumstances (related
>to the type of value added services required & the set of services
>supported by the visited network &  the possibility to transfer a user
>profile--see my previous messages) it is possible to avoid sending
>signalling back to the caller home domain. This applies to calls towards
>the fixed network (which may be routed by TRIP) and for calls to local
>mobiles. For roaming mobiles you fall back to the prvious case: still
>you don' t need to send signalling to the caller home domain, but you
>have a RTT to the callee home domain.



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To: Stephen Terrill <stephen.terrill@era.ericsson.se>
Cc: Ashutosh Dutta <adutta@telcordia.com>, Michael Thomas <mat@cisco.com>,
        James Kempf <James.Kempf@Eng.Sun.COM>, sip@lists.bell-labs.com,
        jdrosen@dynamicsoft.com
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
References: <852568E1.00456034.00@notes949.cc.telcordia.com> <3924E4D8.6D9B92E6@era.ericsson.se>
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Hello Stephen,

Stephen Terrill wrote:

> Hi All,
>
> This is under the assumption that as a user moves during a conversation, the point of presence to the IP also moves.  In the popular radio access systems that I am aware of, this doesn´t happen, the "mico-mobility" (as it is sometimes refered to) is taken care of by the packer radio access network.
>

As you will know, that is only *part* of the picture....

Cheers
Giuseppe


>
> Regards,
>
> ..//Steve
>
> Ashutosh Dutta wrote:
>
> > As far as whether to talk to the home proxy or local proxy, for a wireless
> > roaming user which is  changing domains frequently during mid-session-call and
> > is on the move, it may be better to talk to the local proxy rather than going to
> > the home proxy all the way during every hand-off (for latency reason I guess),
> > of course there needs to be a security association between the local-proxy and
> > home proxy or via public-proxy. This is one of the things we were trying to
> > debate during SIP-2000 conference in Paris last week to see the trade-off
> > between either talking to home-proxy or local domain proxy during frequent
> > hand-off, then there is also a possibility of interaction between local SIP
> > server and AAA server for a roaming user's profile verification during domain
> > handoff.
> >
> > Thanks
> > Ashutosh
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--
----------------------------------------------------------
Giuseppe RICAGNI
System Architecture and Product Planning
Broadband Networks
Italtel
via Reiss Romoli - C10
20019 Castelletto di Settimo Milanese (MILANO)
ITALY

mailto:giuseppe.ricagni@italtel.it
----------------------------------------------------------


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From sip-admin@lists.bell-labs.com  Fri May 19 06:22:24 2000
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Subject: Re: [SIP] Problem with definition of In-Reply-To grammar
To: jdrosen@dynamicsoft.com (Jonathan Rosenberg)
Date: Fri, 19 May 2000 05:21:49 -0500 (CDT)
Cc: schulzrinne@cs.columbia.edu (Henning Schulzrinne),
        gethin@ubiquity.net (Gethin Liddell),
        sip@lists.bell-labs.com (Sip Mail List)
In-Reply-To: <39238002.E53C2169@dynamicsoft.com> from "Jonathan Rosenberg" at May 18, 2000 01:30:42 AM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>> Indeed, earlier we had agreed to use 'token' | 'token' @ 'token' or
>> something like that as the appropriate BNF entity.
>
>I agree completely. There is no reason to make life more complicated
>here.

To keep overaggressive parsers from choking on e.g.
1234@192.168.1.20@domain.com, I'd propose:

token *( "@" token )

...with a RECOMMENDATION that they be of the format currently described
in 2543.

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


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From sip-admin@lists.bell-labs.com  Fri May 19 06:49:16 2000
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Subject: Re: [SIP] Forcing call setup through home proxy (solution?)
To: Jorgen.Bjorkner@hotsip.com (=?iso-8859-1?Q?J=F6rgen_Bj=F6rkner?=)
Date: Fri, 19 May 2000 05:49:02 -0500 (CDT)
Cc: giuseppe.ricagni@italtel.it ('Giuseppe Ricagni '),
        mat@cisco.com ('Michael Thomas '),
        sip@lists.bell-labs.com ('sip@lists.bell-labs.com ')
In-Reply-To: <2160ABC55998D311821900508B2C14BB269D54@SPEED01> from "=?iso-8859-1?Q?J=F6rgen_Bj=F6rkner?=" at May 19, 2000 09:46:17 AM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>As I said before, some services to set up a call might depend on =
>information
>that only is available in the callers home network, and for security or
>policy reasons might be impossible to access from other networks. In =
>this
>case MUST the call setup signaling go through the home network.
>
>I belive that this could be solved by adding a route header field =
>containing
>the address of the home server in the Invite message. If you follow the =
>spec
>must the first proxy in the visited network route the Invite to the =
>home
>proxy. This method allows for optimizing call setup when there is no =
>service
>that requires routing through the home network, and a method to =
>indicate
>when so should be the case.

At the third bakeoff, I went around to a number of people and
asked how their proxies would behave if my client did this sort of
thing. I got a variety of mixed answers, none of which were
very encouraging. Some people wouldn't process it, for example,
unless it had a To: tag (since routes can't appear on leg
setup messages, or so goes the thinking). Most would completely
forgo their internal logic and blindly forward the message
(thus making their purpose only to add to the number of hops
in the call signalling route).

I've been mulling this problem since. I don't think there's
really a good solution; however, I want to run something by
the group.

Back in the day, you could manually route e-mail messages through
a variety of servers by specifying an email address of the format
adam.roach%internal_host@ericsson.com. Upon receiving this,
ericsson.com would turn the last % into an @ and continue
routing the message. This could be indefinitely chained
to produce something like adam.roach%host3%host2%host1@ericsson.com
which would route the message through ericsson.com, then host1,
then host2, and finally deliver it to host3.

So, what if we specified a fairly simple username convention
stating that proxies MAY do something similar; for example,
you could send something like:

INVITE sip:jorgen.bjorkner%40ericsson.com@firewall.mycompany.com SIP/2.0

...to the machine "localservices.mycompany.com". When the
machine "firewall.mycompany.com" received the message, based on
the presence of an escaped @ sign in the user field, he would
strip the machine name in the request URI, expand the rightmost
escaped @, and use that for his new request URI:

INVITE sip:jorgen.bjorkner@ericsson.com SIP/2.0

Questions? Comments? Is there enough interest in this to
warrant putting together a formal document?

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


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From sip-admin@lists.bell-labs.com  Fri May 19 09:13:09 2000
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Cc: giuseppe.ricagni@italtel.it, mat@cisco.com, sip@lists.bell-labs.com
Subject: RE: [SIP] Forcing call setup through home proxy (solution?)
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Can't the Request-URI be used to achieve this functionality? 

UA Oriented:

	I.e. the SIP message's IP destination could be the serving proxy in
the visited network and the Request-URI could indicate the home proxy. This
would allow the UA to be in control over whether the home proxy was in the
loop. See section 6.33.4 in
http://www.cs.columbia.edu/~hgs/sip/drafts/draft-ietf-sip-2543bis-00.pdf. [
Page 51 ]

Script or Network Oriented:

	An network oriented alternative would be to have scripts, profile or
triggers loaded on the serving proxy to rewrite or include the Request-URI
indicating involvement of the home server. This may or may not run into the
"gag me with a spoon" issue that Mike alluded to depending on where the
script came from.

Really Crapy Solution:

	Yet another network oriented method would be to simply have the
service logic on the serving proxy simply proxy the message to the address
of the home proxy; however, this would only work if no other proxies are
needed to be in the loop between the serving and home proxies- don't
recommend. This last one is a bad solution that I personally believe
violates the principles of SIP. It's a hack.

I believe all of these methods except the last would allow a per-message
selection of home proxy involvement, be legal in SIP and not go against the
design principles of the architecture. 

Are there any issues with the SIP violations that anyone is aware of for the
first two methods above?



> -----Original Message-----
> From:	Adam B. Roach [SMTP:Adam.Roach@Ericsson.com]
> Sent:	Friday, May 19, 2000 5:49 AM
> To:	Jorgen.Bjorkner@hotsip.com
> Cc:	giuseppe.ricagni@italtel.it; mat@cisco.com; sip@lists.bell-labs.com
> Subject:	Re: [SIP] Forcing call setup through home proxy (solution?)
> 
> >As I said before, some services to set up a call might depend on =
> >information
> >that only is available in the callers home network, and for security or
> >policy reasons might be impossible to access from other networks. In =
> >this
> >case MUST the call setup signaling go through the home network.
> >
> >I belive that this could be solved by adding a route header field =
> >containing
> >the address of the home server in the Invite message. If you follow the =
> >spec
> >must the first proxy in the visited network route the Invite to the =
> >home
> >proxy. This method allows for optimizing call setup when there is no =
> >service
> >that requires routing through the home network, and a method to =
> >indicate
> >when so should be the case.
> 
> At the third bakeoff, I went around to a number of people and
> asked how their proxies would behave if my client did this sort of
> thing. I got a variety of mixed answers, none of which were
> very encouraging. Some people wouldn't process it, for example,
> unless it had a To: tag (since routes can't appear on leg
> setup messages, or so goes the thinking). Most would completely
> forgo their internal logic and blindly forward the message
> (thus making their purpose only to add to the number of hops
> in the call signalling route).
> 
> I've been mulling this problem since. I don't think there's
> really a good solution; however, I want to run something by
> the group.
> 
> Back in the day, you could manually route e-mail messages through
> a variety of servers by specifying an email address of the format
> adam.roach%internal_host@ericsson.com. Upon receiving this,
> ericsson.com would turn the last % into an @ and continue
> routing the message. This could be indefinitely chained
> to produce something like adam.roach%host3%host2%host1@ericsson.com
> which would route the message through ericsson.com, then host1,
> then host2, and finally deliver it to host3.
> 
> So, what if we specified a fairly simple username convention
> stating that proxies MAY do something similar; for example,
> you could send something like:
> 
> INVITE sip:jorgen.bjorkner%40ericsson.com@firewall.mycompany.com SIP/2.0
> 
> ...to the machine "localservices.mycompany.com". When the
> machine "firewall.mycompany.com" received the message, based on
> the presence of an escaped @ sign in the user field, he would
> strip the machine name in the request URI, expand the rightmost
> escaped @, and use that for his new request URI:
> 
> INVITE sip:jorgen.bjorkner@ericsson.com SIP/2.0
> 
> Questions? Comments? Is there enough interest in this to
> warrant putting together a formal document?
> 
> -- 
> Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS
> L-04
> adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081
> USA
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

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charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2651.65">
<TITLE>RE: [SIP] Forcing call setup through home proxy =
(solution?)</TITLE>
</HEAD>
<BODY>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Can't the =
Request-URI be used to achieve this functionality? </FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">UA Oriented:</FONT>
</P>
<UL>
<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">I.e. the SIP =
message's IP destination could be the serving proxy in the visited =
network and the Request-URI could indicate the home proxy. This would =
allow the UA to be in control over whether the home proxy was in the =
loop. See section 6.33.4 in<U> <A =
HREF=3D"http://www.cs.columbia.edu/~hgs/sip/drafts/draft-ietf-sip-2543bi=
s-00.pdf" =
TARGET=3D"_blank">http://www.cs.columbia.edu/~hgs/sip/drafts/draft-ietf-=
sip-2543bis-00.pdf</A></U>. [ Page 51 ]</FONT></P>
</UL>
<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Script or Network =
Oriented:</FONT>
</P>
<UL>
<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">An network oriented =
alternative would be to have scripts, profile or triggers loaded on the =
serving proxy to rewrite or include the Request-URI indicating =
involvement of the home server. This may or may not run into the =
&quot;gag me with a spoon&quot; issue that Mike alluded to depending on =
where the script came from.</FONT></P>
</UL>
<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Really Crapy =
Solution:</FONT>
</P>
<UL>
<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Yet another network =
oriented method would be to simply have the service logic on the =
serving proxy simply proxy the message to the address of the home =
proxy; however, this would only work if no other proxies are needed to =
be in the loop between the serving and home proxies- don't recommend. =
This last one is a bad solution that I personally believe violates the =
principles of SIP. It's a hack.</FONT></P>
</UL>
<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">I believe all of =
these methods except the last would allow a per-message selection of =
home proxy involvement, be legal in SIP and not go against the design =
principles of the architecture. </FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Are there any issues =
with the SIP violations that anyone is aware of for the first two =
methods above?</FONT>
</P>
<BR>
<BR>
<UL>
<P><FONT SIZE=3D1 FACE=3D"Arial">-----Original Message-----</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">From:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Adam B. Roach =
[SMTP:Adam.Roach@Ericsson.com]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Friday, May 19, 2000 5:49 AM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">Jorgen.Bjorkner@hotsip.com</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Cc:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">giuseppe.ricagni@italtel.it; mat@cisco.com; =
sip@lists.bell-labs.com</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT>=
</B> <FONT SIZE=3D1 FACE=3D"Arial">Re: [SIP] Forcing call setup through =
home proxy (solution?)</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt;As I said before, some services to =
set up a call might depend on =3D</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;information</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;that only is available in the =
callers home network, and for security or</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;policy reasons might be =
impossible to access from other networks. In =3D</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;this</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;case MUST the call setup =
signaling go through the home network.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;I belive that this could be =
solved by adding a route header field =3D</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;containing</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;the address of the home server in =
the Invite message. If you follow the =3D</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;spec</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;must the first proxy in the =
visited network route the Invite to the =3D</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;home</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;proxy. This method allows for =
optimizing call setup when there is no =3D</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;service</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;that requires routing through the =
home network, and a method to =3D</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;indicate</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt;when so should be the =
case.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">At the third bakeoff, I went around to =
a number of people and</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">asked how their proxies would behave =
if my client did this sort of</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">thing. I got a variety of mixed =
answers, none of which were</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">very encouraging. Some people =
wouldn't process it, for example,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">unless it had a To: tag (since routes =
can't appear on leg</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">setup messages, or so goes the =
thinking). Most would completely</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">forgo their internal logic and =
blindly forward the message</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">(thus making their purpose only to =
add to the number of hops</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">in the call signalling route).</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I've been mulling this problem since. =
I don't think there's</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">really a good solution; however, I =
want to run something by</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">the group.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Back in the day, you could manually =
route e-mail messages through</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">a variety of servers by specifying an =
email address of the format</FONT>
<BR><FONT SIZE=3D2 =
FACE=3D"Arial">adam.roach%internal_host@ericsson.com. Upon receiving =
this,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">ericsson.com would turn the last % =
into an @ and continue</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">routing the message. This could be =
indefinitely chained</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">to produce something like =
adam.roach%host3%host2%host1@ericsson.com</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">which would route the message through =
ericsson.com, then host1,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">then host2, and finally deliver it to =
host3.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">So, what if we specified a fairly =
simple username convention</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">stating that proxies MAY do something =
similar; for example,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">you could send something like:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">INVITE =
sip:jorgen.bjorkner%40ericsson.com@firewall.mycompany.com =
SIP/2.0</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">...to the machine =
&quot;localservices.mycompany.com&quot;. When the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">machine =
&quot;firewall.mycompany.com&quot; received the message, based =
on</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">the presence of an escaped @ sign in =
the user field, he would</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">strip the machine name in the request =
URI, expand the rightmost</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">escaped @, and use that for his new =
request URI:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">INVITE =
sip:jorgen.bjorkner@ericsson.com SIP/2.0</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Questions? Comments? Is there enough =
interest in this to</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">warrant putting together a formal =
document?</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">-- </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Adam Roach, Ericsson Inc. |&nbsp; Ph: =
+1 972 583 7594 | 1010 E. Arapaho, MS L-04</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">adam.roach@ericsson.com&nbsp;&nbsp; | =
Fax: +1 972 669 0154 | Richardson, TX 75081 USA</FONT>
</P>
<BR>

<P><FONT SIZE=3D2 =
FACE=3D"Arial">_______________________________________________</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">SIP mailing list</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">SIP@lists.bell-labs.com</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://lists.bell-labs.com/mailman/listinfo/sip" =
TARGET=3D"_blank">http://lists.bell-labs.com/mailman/listinfo/sip</A></F=
ONT></U>
</P>
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From sip-admin@lists.bell-labs.com  Fri May 19 09:38:12 2000
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From: Kevin Summers <Kevin.Summers@ttimail.com>
To: "'Lyndon Ong'" <long@nortelnetworks.com>,
        Kevin Summers <Kevin.Summers@ttimail.com>, sip@lists.bell-labs.com
Subject: RE: [SIP] Payload Boundary of ISUP Object?
Date: Fri, 19 May 2000 08:37:24 -0500
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Yes, I think moving it and clearly stating that it is expected that message
type is where encapsulation starts would be ideal.

-----Original Message-----
From: Lyndon Ong [mailto:long@nortelnetworks.com]
Sent: Thursday, May 18, 2000 10:38 AM
To: Kevin Summers; sip@lists.bell-labs.com
Subject: Re: [SIP] Payload Boundary of ISUP Object?


Kevin,

Actually it mentions this in the Example, we could simply move this up to
Section 3.1 instead.

Cheers,

L. Ong

At 10:26 AM 5/18/2000 -0500, Kevin Summers wrote:
>In draft-ietf-sip-isup-mime-01.txt, it is not explicitly stated where
>encapsulation of an ISUP message should begin. It seems logical to begin
>with the message type (as the example in the draft shows), but nowhere does
>the draft prescribe exactly where to start.
>
>I don't think that this decision should to be left to the implementors --
>perhaps we should strengthen the requirements of how the message is
>encapsulated. 
>Any thoughts?
>
>Kevin Summers
>
>
>message scanned for virus content.
>
>telecom technologies, inc.
>http://www.ttiweb.com
>
>
>
>
>_______________________________________________
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>


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From sip-admin@lists.bell-labs.com  Fri May 19 11:59:25 2000
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Hi,

Just joined this list! Has there been a discussion in this list about using
SCTP as the transport for SIP? I would appreciate it if someone could share
their ideas on this topic.

Thanks in advance.

Ganesh Ramamoorthy




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From sip-admin@lists.bell-labs.com  Fri May 19 13:00:48 2000
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Subject: [SIP] Clarification of 302 handling
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I thought this was pretty cut and dry, but I noticed some odd
behaviour in a SIP client today and after looking through the RFC2543bis
draft, I didn't see this spelled out anywhere.

Is it legal to rewrite the To: field in a subsequent request that resulted
from a 302 response? (Can you replace the To: field with the Contact: 
information from the 302 response?)

In section 7.3.3 (302 Moved Temporarily), it states

"The requesting client SHOULD retry the request at the new address(es)
 given by the Contact: header field."

I interpret this to mean take the same request (To/From/etc. unchanged),
replace the Request-URI with the Contact: in the 302, and re-issue 
the request to the new address specified in the Request-URI.

I can understand changing the From: perhaps, but changing the To: seems 
a bit odd. If you can't count on the To: header being immutable to an
extent, it makes service creation a little more difficult.

Comments?
Sean

--
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From sip-admin@lists.bell-labs.com  Fri May 19 13:03:08 2000
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To: Adam.Roach@Ericsson.com, Jorgen.Bjorkner@hotsip.com,
        gmorrow@nortelnetworks.com
Subject: RE: [SIP] Forcing call setup through home proxy (solution?)
Cc: giuseppe.ricagni@italtel.it, mat@cisco.com, sip@lists.bell-labs.com
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> Can't the Request-URI be used to achieve this functionality? 
>
Yes, for a single hop. If you want to route your messsage through multiple hops,
you can either do what Adam suggested or use Route: headers in the INVITE message. I prefer
the later because it would require the least number of changes to existing proxies. 

> Script or Network Oriented:
> 
> 	An network oriented alternative would be to have scripts, profile or
> triggers loaded on the serving proxy to rewrite or include the Request-URI
> indicating involvement of the home server. This may or may not run into the
> "gag me with a spoon" issue that Mike alluded to depending on where the
> script came from.

This is really horrible, mostly because it is overly complex and buys you nothing over
the previous solution. Also, the concept of using scripts to control this seems a 
dangerous road to go down.

> 
> Really Crapy Solution:
> 
> 	Yet another network oriented method would be to simply have the
> service logic on the serving proxy simply proxy the message to the address
> of the home proxy; however, this would only work if no other proxies are
> needed to be in the loop between the serving and home proxies- don't
> recommend. This last one is a bad solution that I personally believe
> violates the principles of SIP. It's a hack.
>
This is even more complex and still buys you nothing over the first solution.

 
The only reason I would imagine involving the network in this case is if you want the
network to have some say in how the call setup is routed (which of course you might). 
If the (roaming) terminal is intelligent enough to detemine that it wants to route through 
it's home proxy, this is easily done by the terminal.

--
Sean Olson <sean.olson@ericsson.com>




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Igor Slepchin wrote:
> 
> Sounds like the new file still have the old date on it: it's dated April
> 2000...
> 
Thanks for catching that. If you downloaded the draft and it doesn't say
"May 2000", you got the old one. Both PS and PDF versions should now be
correct.


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There is presumably nothing that can keep a user from
cutting-and-pasting the Contact information into a new call To field
(particularly if the UA supports ask-me-before-redirection), but that
would seem to be unusual behavior.

Sean Olson wrote:
> 
> I thought this was pretty cut and dry, but I noticed some odd
> behaviour in a SIP client today and after looking through the RFC2543bis
> draft, I didn't see this spelled out anywhere.
> 
> Is it legal to rewrite the To: field in a subsequent request that resulted
> from a 302 response? (Can you replace the To: field with the Contact:
> information from the 302 response?)
> 
> In section 7.3.3 (302 Moved Temporarily), it states
> 
> "The requesting client SHOULD retry the request at the new address(es)
>  given by the Contact: header field."
> 
> I interpret this to mean take the same request (To/From/etc. unchanged),
> replace the Request-URI with the Contact: in the 302, and re-issue
> the request to the new address specified in the Request-URI.
> 
> I can understand changing the From: perhaps, but changing the To: seems
> a bit odd. If you can't count on the To: header being immutable to an
> extent, it makes service creation a little more difficult.
> 
> Comments?
> Sean
> 
> --
> Sean Olson <sean.olson@ericsson.com>
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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From sip-admin@lists.bell-labs.com  Fri May 19 14:04:23 2000
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Adam B. Roach wrote:
> 
> >> Indeed, earlier we had agreed to use 'token' | 'token' @ 'token' or
> >> something like that as the appropriate BNF entity.
> >
> >I agree completely. There is no reason to make life more complicated
> >here.
> 
> To keep overaggressive parsers from choking on e.g.
> 1234@192.168.1.20@domain.com, I'd propose:
> 
> token *( "@" token )
> 
> ...with a RECOMMENDATION that they be of the format currently described
> in 2543.
> 

This wouldn't be backward-compatible. Why encourage such 'strange'
formats?


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From sip-admin@lists.bell-labs.com  Fri May 19 14:06:52 2000
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Right. Any one of several possible behaviors are allowed and will work:

1. The UA basically changes the request URI and uses a different branch
ID in the top Via, but the CSeq, To, From and Call-ID are the same.
2. The UA uses a new To field, request URI, but the Call-ID is unchanged
(doesn't matter if the CSeq changes, since its defined within the
To/From/Call-ID space)
3. The UA uses a brand new To, From, Call-ID, CSeq, etc. as if it were a
totally new request.

Probably (1) is best, since it will be noted by proxies as being the
same request transaction, and thus avoid contacting locations already
tried. All three work, but I wouldn't recommend (2), since this might
introduce interactions with multiparty conferencing stuff.

-Jonathan R.

Henning Schulzrinne wrote:
> 
> There is presumably nothing that can keep a user from
> cutting-and-pasting the Contact information into a new call To field
> (particularly if the UA supports ask-me-before-redirection), but that
> would seem to be unusual behavior.
> 
> Sean Olson wrote:
> >
> > I thought this was pretty cut and dry, but I noticed some odd
> > behaviour in a SIP client today and after looking through the RFC2543bis
> > draft, I didn't see this spelled out anywhere.
> >
> > Is it legal to rewrite the To: field in a subsequent request that resulted
> > from a 302 response? (Can you replace the To: field with the Contact:
> > information from the 302 response?)
> >
> > In section 7.3.3 (302 Moved Temporarily), it states
> >
> > "The requesting client SHOULD retry the request at the new address(es)
> >  given by the Contact: header field."
> >
> > I interpret this to mean take the same request (To/From/etc. unchanged),
> > replace the Request-URI with the Contact: in the 302, and re-issue
> > the request to the new address specified in the Request-URI.
> >
> > I can understand changing the From: perhaps, but changing the To: seems
> > a bit odd. If you can't count on the To: header being immutable to an
> > extent, it makes service creation a little more difficult.
> >
> > Comments?
> > Sean
> >
> > --
> > Sean Olson <sean.olson@ericsson.com>
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Fri May 19 14:37:54 2000
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Jonathan Rosenberg wrote:
> 
> Right. Any one of several possible behaviors are allowed and will work:
> 
> 1. The UA basically changes the request URI and uses a different branch
> ID in the top Via, but the CSeq, To, From and Call-ID are the same.
> 2. The UA uses a new To field, request URI, but the Call-ID is unchanged
> (doesn't matter if the CSeq changes, since its defined within the
> To/From/Call-ID space)
> 3. The UA uses a brand new To, From, Call-ID, CSeq, etc. as if it were a
> totally new request.
> 
> Probably (1) is best, since it will be noted by proxies as being the
> same request transaction, and thus avoid contacting locations already
> tried. All three work, but I wouldn't recommend (2), since this might
> introduce interactions with multiparty conferencing stuff.
> 
Wording added:

The new request can take two different forms.  In the first approach,   
the \header{To}, \header{From}, \header{CSeq} and \header{Call-ID}      
header fields in the new request are the same as in the original        
request, with a new \header{branch} identifier in the \header{Via}      
header field.  Proxies {\SHOULD} follow this behavior and UACs {\MAY}.  
UAs {\MAY} also use the \header{Contact} information for the \header{To}
header field, as well as a new \header{Call-ID} value.


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From sip-admin@lists.bell-labs.com  Fri May 19 14:46:52 2000
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        Sean Olson <eussean@exu.ericsson.se>, sip@lists.bell-labs.com
Subject: Re: [SIP] Clarification of 302 handling
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I believe UAC should increase CSeq if it decides to keep the original
To, From and Call-ID in the new request.

---
Igor Slepchin


Henning Schulzrinne wrote:
> Wording added:
> 
> The new request can take two different forms.  In the first approach,
> the \header{To}, \header{From}, \header{CSeq} and \header{Call-ID}
> header fields in the new request are the same as in the original
> request, with a new \header{branch} identifier in the \header{Via}
> header field.  Proxies {\SHOULD} follow this behavior and UACs {\MAY}.
> UAs {\MAY} also use the \header{Contact} information for the \header{To}
> header field, as well as a new \header{Call-ID} value.
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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Giuseppe Ricagni wrote:
> 
> > This presumes that there aren't any other proxies which decide to
> > record-route. When a call is made, it will generally pass through
> > many
> > proxies, each providing differing levels of service. Some of these
> > will
> > want to record-route.I don't want to prevent them from doing so.
> 
> Basically, the only problem here is that the callee is roaming.
> 
> As Henning reminded us, this requires a database (let' s call it Home
> Subscriber Server: HSS, a kind of GSM HLR) lookup by the "calling
> proxy" in the callee home domain, to have information about the
> address of the proxy the callee is registered at.

In SIP, we call that database DNS.

> 
> This lookup can be performed directly by the caller proxy, or via a
> callee home domain "home proxy".

No - how would we get to the proxy of the callee in the first place
without having identified this server?

> 
> This  "home proxy" is the only proxy you would like to write in the
> record-route and I would like not to.

You are missing my point. You are assuming a very specific configuration
of two proxies, without any services. In real networks, there will be
lots of proxies, many of which are providing a variety of different
services. It is the right of these proxies to record-route in order to
provide those services. I don't know what those services will be, nor do
I know how many proxies there will be. I have heard of trial SIP
networks with six proxies on establishment paths. The great thing about
SIP is that it allows these kinds of things. 

So, my point is that GIVEN that there will be other proxies on the path
for a variety of reasons, you really won't be able to force signaling to
go directly between users, or between users and a local proxy, in most
of the cases.

> 
> IMHO, since  from a technical point of view you could query the HSS
> directly from the calling proxy, you are to involving the "home proxy"
> only for "political reasons", to some extent to "filter" queries to
> the HSS.

No. The home proxy is providing services.

> 
> There is therefore no point in having  such proxy in the record-route.

I doubt you will be able to dictate to service providers that they are
not allowed to provide services in their proxies.

> 
> Also remember we are talking here about basic calls, requesting no
> additional services, or services that   are user profile independent.
> More complex scenarios can be achieved, as I suggested in a previous
> posting:

You can't design protocol mechanisms which do not consider the variety
of cases that might exist.


> With respect to the method used to classify service request within the
> four cathegories, a
> possibility is that  the home domain, at registration time, sends
> information to the visited domain
> on the requirements to meet  to locally execute a given set of
> services for that specific user. If
> the visited domain supports such requirements ==> those services can
> be executed locally.

This seems dubious. It assumes some kind of simple classification of
services which is unlikely to exist, and it also assumes that there is
some kind of transfer of service logic from one domain to another. I
think we have already discussed that this is really far fetched, both
technically and politically.


> 
> > Optimizing a system for the special case where (1) both caller and
> > called party have home domains that are far, (2) they are both
> > roaming
> > in visiting domains that are close, (3) there are no other proxies
> > on
> > the path which record-route, seems like you are optimizing for the
> > wrong
> > case.
> 
> First of all: we are not "optimizing" a system, here: we are adding a
> feature, that can be useful to "optimize" a given set of cases.
> 
> Let's make a simpler  (and very common) example:
> an Italian mobile roaming to Australia, that wants to call a (local !)
> Taxy: is this the wrong case ? (In that case I'm wrong myself, because
> I did it in Adelaide!!!  :-))

This is a good example. I have no problem if, when a user moves
somewhere else, the user decides to select a new ASP for providing
certain services. This ASP has its own proxies and application servers,
which may or may not be physically local to the user. 

The choice is ultimately the end users. My point is that I disagree with
forcing users to obtain services through a certain proxy just because
that service provider is also providing them with their IP level
mobility services. I have no problems allowing users to choose different
service providers at will, including the one providing the IP mobility
services, if they have something compelling to offer, such as the
service you desribe above (basically, a dial plan service).


> > This need not be the case. In a scenario
> > with a smoother handover, I can imagine a brief period where the
> > roaming
> > mobile has both addresses - one in the old domain, one in the new.
> > When
> > it obtains the new one, it does a re-INVITE to *add* a new media
> > stream
> > targeted to the new address. This means it will get both stream of
> > packets during this transition period. Once these new packets start
> > arriving, it can do a re-INVITE again, turning off the stream
> > targeted
> > to the old address. It can then release the old address.
> 
> Sorry, but I think you missed the target.
> 
> The problem is only related to the in-flight packets.
> If I' m in the position of keeping the two address active for a while,
> I will get the in-flight packets on the old
> address anyway !!!! I don' t need any duplicte invite !!!

You would if usage of that extra old address was costing you money, or
it would cease to be possible to use it after some time. As I said, I am
not sure if this is how IP mobility handover is accomplished.

> 
> The problem comes when I lose the old address and there are some
> already in-flight packets addressed to it.
> The number of in-flight packets is proportional to the time needed to
> make  the other endpoint aware of the new address.
> No duplicate Invites or whatever can help.

At the risk of causing even more controversy and flames, I'll note that
this problem is occuring because you are using an application layer
mobility service for a network layer mobility need. If you were using an
IP layer mobility service which allowed you to tell the network which
owned your old address that you have moved, and thus to forward packets
to your new address, you would not have this problem.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Subject: Re: [SIP] Multimedia Conferencing using SIP
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Jonathan Rosenberg wrote:

> No extensions needed for central conference servers. It works just like
> it does in the PSTN - you dial a number which corresponds to a port on a
> bridge. In the case of SIP, you would INVITE to a URL corresponding to a
> port on the conference server, like sip:conference33287372@mcu.com. The
> result appears as a point to point SIP call as far as  your UA is
> concerned, but the server is mixing media and sending the mixed media to
> your UA. You will learn the identity of the other participants through
> RTCP CSRC.
>
> How do you know the right URL to send an INVITE to? Well, how do you
> know what number to dial for a conference in PSTN? You're told out of
> band. URLs can be included in emails, web pages, whatever.
>

In "SIP Call Control Services," the underlying model for establishing a conference seems to be that it is the responsibility of the invited user to send INVITE requests to the existing conference participants.  That is, in a conference that is comprised of User A, B, and C, say A wishes to add D to the conference.  According to the "SIP Call Control Services," A invites D, who, in turn, sends INVITE requests to B and C.
Why is this model preferred over another model, in which the existing conference participants send INVITE requests to the invited?

--Hyong Shim



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Sean Olson wrote:
> 
> > Can't the Request-URI be used to achieve this functionality?
> >
> Yes, for a single hop. If you want to route your messsage through multiple hops,
> you can either do what Adam suggested or use Route: headers in the INVITE message. I prefer
> the later because it would require the least number of changes to existing proxies.

Correct. If you send a request to any proxy, where the request-URI does
not match the domain of the proxy, that proxy is supposed to forward the
request to the proxy indicated in the request URI. In fact, that proxy
could also forward, through its own configuration, to another proxy,
which could then forward to the proxy in the request URI. But, if you
want the UAC to specify multiple hops, use Route header.

Adam wrote:
> At the third bakeoff, I went around to a number of people and
> asked how their proxies would behave if my client did this sort of
> thing. I got a variety of mixed answers, none of which were
> very encouraging. Some people wouldn't process it, for example,
> unless it had a To: tag (since routes can't appear on leg
> setup messages, or so goes the thinking). Most would completely
> forgo their internal logic and blindly forward the message
> (thus making their purpose only to add to the number of hops
> in the call signalling route).

Checking for tags and only obeying Route if there is a tag is probably
not a good idea. It was not meant for this purpose. I think what is
desired by that hack is to ensure that a Route is being followed only if
it was one created by the proxy in a Record-Route. We had discussed, at
the bakeoff, some kind of opaque token which would need to be returned
along with the Route headers, which the proxy could use to confirm this.
I believe that is a much better approach than checking for the tag.

I'll note that verifying that the Route header is the result of
insertion of Record-Route is at odds with Jorgens suggestion of using
Route to force traversal through proxies for purpose of services. I
believe that there is no way to reconcile these at the protocol level;
it is a matter of local policy. Just like some MTAs don't support mail
relay, and others do, so it is with obeying Route headers inserted by a
UA without having seen a Record-Route first. In Jorgens example, this
local proxy is presumably aware that it is there just because of
mobility, and that users will want to get to their home proxies, and
thus obey the Route header even if its in a new request.

Adam later wrote:
> So, what if we specified a fairly simple username convention
> stating that proxies MAY do something similar; for example,
> you could send something like:
> 
> INVITE sip:jorgen.bjorkner%40ericsson.com@firewall.mycompany.com SIP/2.0

I don't see what this buys you above using Route. You are basically
doing exactly the same thing as Route, but gluing the route headers
together through URL encoding and then putting them in the request URI.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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"Vijay K. Gurbani" wrote:
> 
> Jonathan/Henning:
> 
> In RFC2543bis, section 2, 2nd last paragraph on page 17, you say:
> 
>    "The user parameter value "np-queried" indicates that the user part
>     contains a telephone number AND that the number reflects the result
>     of a query to the local number portability database".
> 
> However, in Figure 3 on page 18, the BNF for user-param is:
> 
>    user-param = "user="("phone" | "ip")
> 
> Should it be modified to:
> 
>    user-param = "user="(("phone" | "np-queried") | "ip")
> 
> A look at the archives indicate that Jonathan and Marc Petit-Huguenin
> had a thread on this around May 10, 2000 with Jonathan stating that
> there was no consensus on if this was the right way to do this.  Any
> updates?


None I've seen. I have no problem with the idea of saying that some kind
of query has been done, but this particular solution is very specific to
LNP, and I think there is a bigger picture. 

I would argue that the need for this thing at all arises because the
telephone network uses equivalently formatted numbers to refer to two
things - names and actual routing numbers (i.e., addresses). Its sort of
like what would have happened if IP addresses and domain names were the
same in syntax and formatting, such that you couldn't tell one from the
other without checking with some database. The purpose of LNP is much
like DNS, to translate an abstract name to a real routable address. Now,
in the world of URLs, the namespace (or abstractly, the address space)
of the content of the URL is defined by the scheme. If we view "numbers
as names" and "numbers as addresses" as difference "namespaces", the
theoretically correct way to indicate that LNP has been done is by
defining a new URL scheme for the two:

tel:18005551212 and
rn:17325551212

where LNP (or 800 number lookup in this case) is used to translate.

I am not proposing to do this, since practically speaking, its a pain.
But I'd like something more generic to indicate the namespace that the
number fits into. I don't exactly know what or how to do this. Perhaps
this is really just phone-context, with the context being
"routing-number" once translated? 

-Jonathan R.
> 
> Thanks,
> 
> - vijay
> --
> Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
> IN Architecture and Internet Software Group
> Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
> Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Michael Thomas wrote:
> 
>  >    * When the caller is roaming, under certain circumstances (related to the type of value added
>  >      services required & the set of services supported by the visited network &  the possibility to
>  >      transfer a user profile--see my previous messages)
> 
>    I don't know what services the roaming network
>    wants to provide -- other than trying to shoehorn
>    themselves into the call. Even if, there is nothing
>    to prevent me from obtaining those service even if
>    it's routed through my home proxy first.

There are two different cases - one where the proxy is asking to be on
the call path for an existing call, and the other, where the proxy
wishes to be the local outbound (or inbound) proxy for outgoing (or
incoming) calls. I agree totally that the former - shoehorning into an
existing call - makes little sense technically. For the latter case, I
think the roaming proxy is simply another ASP. I can see reasons why I
might want to use them for some calls, and not for others. I see no
reason to prevent a user from using those proxies, but I don't see a
reason for forcing them to use them.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Henning Schulzrinne wrote:
> 
> Wording added:
> 
> The new request can take two different forms.  In the first approach,
> the \header{To}, \header{From}, \header{CSeq} and \header{Call-ID}
> header fields in the new request are the same as in the original
> request, with a new \header{branch} identifier in the \header{Via}
> header field. 

, and the new address in the \header{Request-URI}.

> Proxies {\SHOULD} follow this behavior and UACs {\MAY}.

\MUST follow this behavior (they can't change the Call-ID, To, From or
CSeq, not any other choice then)


-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Fri May 19 16:24:34 2000
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Subject: Re: [SIP] Clarification of 302 handling
References: <200005191700.MAA21271@b04a45.exu.ericsson.se> <39257C47.75917023@cs.columbia.edu> <39258526.882F0626@dynamicsoft.com> <39258427.FA3783F7@cs.columbia.edu> <39258B9C.4D96C8B@dynamicsoft.com>
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Not neccesarily. No reason that the UA can't do exactly the same as a
proxy does. I argued in a previous email that there are advantages to
this, since it means as the new request is forwarded, proxies can
recognize this as the same transaction and not forward to addresses they
already tried.

-Jonathan R.

Igor Slepchin wrote:
> 
> I believe UAC should increase CSeq if it decides to keep the original
> To, From and Call-ID in the new request.
> 
> ---
> Igor Slepchin
> 
> Henning Schulzrinne wrote:
> > Wording added:
> >
> > The new request can take two different forms.  In the first approach,
> > the \header{To}, \header{From}, \header{CSeq} and \header{Call-ID}
> > header fields in the new request are the same as in the original
> > request, with a new \header{branch} identifier in the \header{Via}
> > header field.  Proxies {\SHOULD} follow this behavior and UACs {\MAY}.
> > UAs {\MAY} also use the \header{Contact} information for the \header{To}
> > header field, as well as a new \header{Call-ID} value.
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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        sip@lists.research.bell-labs.com,
        "'Mohammad.Vakil@outreachtech.com'" <Mohammad.Vakil@outreachtech.com>
Subject: Re: [SIP] Multimedia Conferencing using SIP
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Hyong Sop Shim wrote:
> 
> Jonathan Rosenberg wrote:
> 
> > No extensions needed for central conference servers. It works just like
> > it does in the PSTN - you dial a number which corresponds to a port on a
> > bridge. In the case of SIP, you would INVITE to a URL corresponding to a
> > port on the conference server, like sip:conference33287372@mcu.com. The
> > result appears as a point to point SIP call as far as  your UA is
> > concerned, but the server is mixing media and sending the mixed media to
> > your UA. You will learn the identity of the other participants through
> > RTCP CSRC.
> >
> > How do you know the right URL to send an INVITE to? Well, how do you
> > know what number to dial for a conference in PSTN? You're told out of
> > band. URLs can be included in emails, web pages, whatever.
> >
> 
> In "SIP Call Control Services," the underlying model for establishing a conference seems to be that it is the responsibility of the invited user to send INVITE requests to the existing conference participants.


That is one model, but not the only one SIP provides. SIP has four ways
to do multiparty conferencing:

1. dialup conference bridge: call the bridge just like you call a
person. Conference is identified by request URI. Works with rfc2543 - no
extensions needed. 
2. distributed multiparty conferencing - no server. Fully distributed.
This is what is described in the now-expired draft, and is work in
progress.
3. multicast conferences - you can run your conference on multicast.
Simply INVITE the person to join the multicast session. Works with
baseline SIP. In fact, this was the initial purpose of SIP. In this
case, there is not a full mesh of SIP signaling.
4. 3-way with local MC/MP function: A and B are talking. A wishes to add
C to the call. A can simply call C, but also act as a mixer, so that the
media it sends to B contains the A+C media, and the media to C is the
A+B media. RTCP CSRC indicate who is in the conference. This also works
with baseline SIP. It imposes additional burden on the UA, but otherwise
provides this standard feature in a simple way. 



There is no preferred way. It depends on the particular application. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Subject: Re: [SIP] Clarification of 302 handling
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If a UA does this, any proxy compliant to RFC2543 will consider the new
request a retransmission of the old one. Top Via is not part of the call
leg (aka transaction) id according to the RFC. Besides, I see absolutely
no point in making the UAC keep the Cseq the same: any proxy interested
in following call progress should be able to match the re-INVITEs with
increased CSeq anyway so why not increase the CSeq and make everything
consistent?

---
Igor Slepchin


Jonathan Rosenberg wrote:
> 
> Not neccesarily. No reason that the UA can't do exactly the same as a
> proxy does. I argued in a previous email that there are advantages to
> this, since it means as the new request is forwarded, proxies can
> recognize this as the same transaction and not forward to addresses they
> already tried.
> 
> -Jonathan R.
> 
> Igor Slepchin wrote:
> >
> > I believe UAC should increase CSeq if it decides to keep the original
> > To, From and Call-ID in the new request.
> >
> > ---
> > Igor Slepchin
> >
> > Henning Schulzrinne wrote:
> > > Wording added:
> > >
> > > The new request can take two different forms.  In the first approach,
> > > the \header{To}, \header{From}, \header{CSeq} and \header{Call-ID}
> > > header fields in the new request are the same as in the original
> > > request, with a new \header{branch} identifier in the \header{Via}
> > > header field.  Proxies {\SHOULD} follow this behavior and UACs {\MAY}.
> > > UAs {\MAY} also use the \header{Contact} information for the \header{To}
> > > header field, as well as a new \header{Call-ID} value.
> > >
> > > _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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Igor Slepchin wrote:
> 
> If a UA does this, any proxy compliant to RFC2543 will consider the new
> request a retransmission of the old one. Top Via is not part of the call
> leg (aka transaction) id according to the RFC.

Its only consider a retransmission if the request URI is the same
(recall the new rules for loop detection). If the request URI is the
same, then you get exactly what you want - the proxy treats this as a
retransmission  or rejects it as a merged request. We've already tried
that address.

 Besides, I see absolutely
> no point in making the UAC keep the Cseq the same: any proxy interested
> in following call progress should be able to match the re-INVITEs with
> increased CSeq anyway so why not increase the CSeq and make everything
> consistent?

I am not advocating using the same CSeq for purposes of tracking calls,
but for purposes of unifying the way a proxy does recursion and the way
a UAC does recursion. You get the benefit of not trying places again
you've already tried.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Fri May 19 17:13:04 2000
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Subject: Re: [SIP] Question abt. Content-Disposition
References: <652568E4.001C1EAC.00@sampark.hss.hns.com>
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The purpose of this is that we have found that Content-Type defines the
*syntax* of the content, but not the semantics. In some cases, one
implies the other, and in other cases, they don't. As an example, if a
receive an INVITE with a multipart, and one of the multiparts contains a
URI list, what is that URI list for? Is it content to be displayed (like
a jpeg), or is it a way to fetch the session description? The idea with
Content-Disposition is that it tells you the purpose of the content (and
also if its optional or mandatory).

Another application of it is in:
http://search.ietf.org/internet-drafts/draft-lennox-sip-reg-payload-00.txt

which calls it Content-Purpose, but the need is the same - what is the
purpose of the payload in the registration message.

Some of the possible purposes include:

session (like SDP)
inline (display to the user, like JPEG or html)
authorization (like an OSP token)
sip-cgi (a CGI script)
cpl (A CPL script)



-Jonathan R.

archow@hss.hns.com wrote:
> 
> Hi,
> 
> According to the description of Content-Dispotion (C-D) header, its role is
> to tell the UA whether the message (part) has to be
> necessarily idenfied or not by the UA.
> 
> So, if I had a MIME header with 2 parts, with my first part being an ISUP
> encoded payload and my second part being a message which I really dont care
> if the UA understands or not,  I do:
> 
> --- begin ----
> INVITE sip:01628434456@dcs1.nortelnetworks.com SIP/2.0
> <etc ... etc>
> Content-Type: multipart/mixed; boundary=unique-boundary-1
> 
> --unique-boundary-1
> Content-type:application/isup
> Content-Transfer-Encoding: binary
> 
> .. <binary payload > ...
> 
> --unique-boundary-1
> Content-type:application/moohaa
> Content-Diposition:inline;handling=optional
> 
> I dont really care if you understand me or not
> 
> --unique-boundary-1--
> 
> --- end ---
> 
> Is the above correct ?
> 
> I could find any reference to what the disposition type token is all abt,
> and when I should specify which.
> Also, what is the extension-param in disposition type for ?
> 
> If all its use is to tell the UA to mandatorily/optionally understand a
> message part, then isnt
> 
> Content-Disposition - "Content-Disposition" ":" dispositon-type
> disposition-type = "optional" | "required"
> enough ?
> Regds
> Arjun
> 
> --
> Arjun Roychowdhury @ Hughes Software Systems
> 
> _______________________________________________
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Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
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From sip-admin@lists.bell-labs.com  Fri May 19 17:32:31 2000
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From: "Schroeder, Tim" <schroeder@tri.sbc.com>
To: "'Lars Berggren'" <lars@intertex.se>, sip@lists.bell-labs.com
Subject: RE: [SIP] Comments on Session Timer draft
Date: Fri, 19 May 2000 16:31:42 -0500
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The session timer draft (draft-ietf-sip-session-timer-01.txt) as written is
asymmetric, in that it seems to assume that the UAC is more likely to
impolitely abandon the session, and that the proxies and UAS need some way
to recover from that.

I'd like to see the reverse case addressed as well: where the UAC (or a
proxy) is distrustful of the UAS, and needs some way to clean up if the UAS
impolitely abandons the session.

I suppose the keep-alive re-INVITES can still be sent by the UAC, and it
will be the lack of a response (rather than the lack of an incoming message)
which informs the UAC and the proxies that the UAS has dropped off.

But in regard to the Open Issues list, I think the answer to the second
issue (should we allow the UAS to insert a Require: timer header?) should be
YES.  Just like the UAS can reject a call that won't be set up with a
session timer, the UAC should be able to avoid having a call set up without
a session timer.  It would do this by using the "Require: timer" header
rather than the weaker "Supported: timer" header.  If the UAS cannot handle
the feature, the call would be rejected, rather than set up without the
(demanded) timer feature.

And for the third Open Issue on the list, a proxy should also be able to add
the "Require: timer" header if the UAC merely "supports" but does not
"require" the feature.  Just like the proxy can tighten up the timing
requirements by lowering the expiration time, it should be able to tighten
up the requirements by making the feature "required" rather than merely
"supported".  Then, if the UAS did not implement that feature, the request
would be denied.  [This might take some more thought; it would seem odd to
the UAC for its request to end up denied for lack of a feature it did not
require.]

Alternatively, maybe there's no support needed in the UAS at all: the
re-INVITES are all sent from the UAC, and the UAS needn't know anything
about the feature to process them.  This is similar to what Lars Berggren  <
<mailto:lars.berggren@intertex.se> mailto:lars.berggren@intertex.se>
proposes:

>>
For the Session Timer to be applied for a call leg, the specifications in
the draft requires that both the UAC and UAS understand the extension
(See example 8.5). However, since the keep-alive mechanism is achieved by
sending 'standard' re-INVITEs, it is sufficient (and useful) that at least
one of the parties is aware of the Session Timer to accomplish a
keep-alive mechanism. The usefulness of such an approach is for example
when a UAC supports 'timer', a stateful proxy server, SPS, wants to apply
a Session Timer to the call and the UAS doesn't understand the extension.

Here, the call will be setup without Session Timer even though the SPS
wanted one and the UAC could do the necessary re-inviting. The UAS doesn't
need to know about Session Timers in order to process re-INVITEs properly.
The only thing that forces that the UAS have to be aware of the Session
Timer is the statement in the draft that the UAS SHOULD send BYE if no
re-INVITE has arrived when the session expires. Well, the UAC could do the
sending of the BYE if it does not intend to update the timer.

So, to solve this problem, I would like to propose that the SPS could add
a Session-Expires header in the response if the SPS knows the UAC supports
'timer' and the response does not contain a Session-Expires header.
<<
 
At any rate, I'd like the draft to address the cases where any of the
UAC/UAS/proxy *require* the session timer feature.  I'd consider it OK for
the session to not get set up if the session timer feature weren't supported
by all (or enough) of the UAs, but not OK for the session to get set up
without a timer if one were required anywhere along the path.
 
Also, the third to the last paragraph in section 4 (beginning "A UAC may use
the refreshing re-INVITE as a normal SIP re-INVITE ...") makes no sense to
me.  It talks about sending an *updated* session description and using the
origin field to indicate the description is *unchanged*.

Tim Schroeder
tschroeder@tri.sbc.com
SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759

 


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From sip-admin@lists.bell-labs.com  Fri May 19 18:36:26 2000
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From: "Dean Willis" <dean.willis@softarmor.com>
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Subject: RE: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task force in Adelaide)
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Ah. The networkstatus-aware proxy is effectively advising the SIP user of congestion in the DiffServ network. Kind of like those radio announcers that say "Highway 18 is shut down again -- please use surface streets or just stay home".

The real question is to define a mechanism that allows the health of the network to be assessed in some sort of meaningful way so that this knowledge may be made available to the UAC.

--
Dean



> -----Original Message-----
> From: Michael Thomas [mailto:mat@cisco.com]
> Sent: Thursday, May 18, 2000 4:32 PM
> To: Dean Willis
> Cc: IETF SIP; Michael Thomas
> Subject: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task force
> in Adelaide)
> 
> 
> Dean Willis writes:
>  > Consider it in an aggregation model, like a donut. The chewy 
> outside parts of the donut are controlled by IntServ. Traffic 
> leaps across the DiffServ donut-hole only if the heuristic 
> process thinks (IE, is reasonably sure) that there is adequate bandwidth.
> 
>    OK, but I still don't see what that has to do
>    with proxies. That sounds like something an
>    edge router would do before it admits diffserv
>    traffic into the diffserv core. Proxies are even
>    more clueless because they never see the actual
>    traffic patterns.
> 
> 		Mike
> ÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÒ ùšŠX§‚X¬µ"þX¬¶ÏÛzYÿ•¦ìýÊ&†ÛiÿÿåŠËlý·¥—ùZnÏÜ¢oæj)fjåŠËbú?


From sip-admin@lists.bell-labs.com  Sat May 20 09:10:28 2000
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From: "Sinnreich, Henry" <Henry.Sinnreich@WCOM.Com>
Subject: RE: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task for	ce
 in Adelaide)
To: "'Dean Willis'" <dean.willis@softarmor.com>,
        Michael Thomas <mat@cisco.com>
Cc: IETF SIP <sip@lists.bell-labs.com>
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>The real question is to define a mechanism that allows the 
>health of the network to be assessed in some sort of 
>meaningful way so that this knowledge may be made available to the UAC.

The model in <sinnreich-sip-qos-osp> actually deals with this problem.
Upon failure of the RSVP request, the telephony endpoint can alert the user:
"QoS cannot be provided, would you like to proceed with the call".

The awareness of the originating SIP server, though we not explicitly
discussed is can also be dealt with by the model.

We have shown that the transit network providing DiffServ to the edge
networks have to police the ingress traffic, to make sure that that the
premium QoS traffic injected does not exceed the SLS. This job is done by
the border routers in the transit networks.

The border router can alert the edge router in the access network when
excess traffic is being passed, and this could be also used to alert the
bandwidth and policy manager in the access network. The policy manager will
refuse to authorize new requests for QoS enabled voice calls coming from
application servers, including the SIP server. The SIP server will therefore
know...

Please let me know if there are any weaknesses in the model, as shown in the
diagrams and call flows.
http://ietf.org/internet-drafts/draft-sinnreich-interdomain-sip-qos-osp-01.t
xt

Henry

>-----Original Message-----
>From: Dean Willis [mailto:dean.willis@softarmor.com]
>Sent: Friday, May 19, 2000 5:35 PM
>To: Michael Thomas
>Cc: IETF SIP
>Subject: RE: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task
>force in Adelaide)
>
>
>
>Ah. The networkstatus-aware proxy is effectively advising the 
>SIP user of congestion in the DiffServ network. Kind of like 
>those radio announcers that say "Highway 18 is shut down again 
>-- please use surface streets or just stay home".
>
>The real question is to define a mechanism that allows the 
>health of the network to be assessed in some sort of 
>meaningful way so that this knowledge may be made available to the UAC.
>
>--
>Dean
>
>
>
>> -----Original Message-----
>> From: Michael Thomas [mailto:mat@cisco.com]
>> Sent: Thursday, May 18, 2000 4:32 PM
>> To: Dean Willis
>> Cc: IETF SIP; Michael Thomas
>> Subject: SIP DiffServe (was RE: [SIP] Minutes of SIP 
>Security task force
>> in Adelaide)
>> 
>> 
>> Dean Willis writes:
>>  > Consider it in an aggregation model, like a donut. The chewy 
>> outside parts of the donut are controlled by IntServ. Traffic 
>> leaps across the DiffServ donut-hole only if the heuristic 
>> process thinks (IE, is reasonably sure) that there is 
>adequate bandwidth.
>> 
>>    OK, but I still don't see what that has to do
>>    with proxies. That sounds like something an
>>    edge router would do before it admits diffserv
>>    traffic into the diffserv core. Proxies are even
>>    more clueless because they never see the actual
>>    traffic patterns.
>> 
>> 		Mike
>> Hfæj)bz	b²Ô^>X¬¶ÆÞ-YZnÇ(sm§ÿåSËlmée*¦ìr?¿(tm)¨¥(tm)©ÿ-+-SwèþÈ©
>


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From sip-admin@lists.bell-labs.com  Sat May 20 09:26:26 2000
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From: "Sinnreich, Henry" <Henry.Sinnreich@WCOM.Com>
Subject: FW: SIP DiffServ (was RE: [SIP] Minutes of SIP Security task for	ce in
 Adelaide)
To: IETF SIP <sip@lists.bell-labs.com>, Michael Thomas <mat@cisco.com>,
        "'Dean Willis'" <dean.willis@softarmor.com>
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Sorry for the resend - there were typos. Henry

>The real question is to define a mechanism that allows the 
>health of the network to be assessed in some sort of 
>meaningful way so that this knowledge may be made available to the UAC.

The model in <sinnreich-sip-qos-osp> actually deals with this problem.
Upon failure of the RSVP request, the telephony endpoint can alert the user:
"QoS cannot be provided, would you like to proceed with the call?".

The awareness of the originating SIP server, though not explicitly
discussed, can also be dealt with by the model.

We have shown that the transit networks providing DiffServ to the edge
networks, have to police the ingress traffic, to make sure that the
premium QoS traffic injected does not exceed the SLS. This job is done by
the border routers in the transit networks.

The border router can alert the edge router in the access network when
excess traffic is being passed, and this could be also used to alert the
bandwidth and policy manager in the access network. The policy manager will
refuse to authorize new requests for QoS enabled voice calls coming from
application servers, including the SIP server. The SIP server will therefore
know...

Please let me know if there are any weaknesses in the model, as shown in the
diagrams and call flows.
http://ietf.org/internet-drafts/draft-sinnreich-interdomain-sip-qos-osp-01.t

Henry

>-----Original Message-----
>From: Dean Willis [mailto:dean.willis@softarmor.com]
>Sent: Friday, May 19, 2000 5:35 PM
>To: Michael Thomas
>Cc: IETF SIP
>Subject: RE: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task
>force in Adelaide)
>
>
>
>Ah. The networkstatus-aware proxy is effectively advising the 
>SIP user of congestion in the DiffServ network. Kind of like 
>those radio announcers that say "Highway 18 is shut down again 
>-- please use surface streets or just stay home".
>
>The real question is to define a mechanism that allows the 
>health of the network to be assessed in some sort of 
>meaningful way so that this knowledge may be made available to the UAC.
>
>--
>Dean
>
>
>
>> -----Original Message-----
>> From: Michael Thomas [mailto:mat@cisco.com]
>> Sent: Thursday, May 18, 2000 4:32 PM
>> To: Dean Willis
>> Cc: IETF SIP; Michael Thomas
>> Subject: SIP DiffServe (was RE: [SIP] Minutes of SIP 
>Security task force
>> in Adelaide)
>> 
>> 
>> Dean Willis writes:
>>  > Consider it in an aggregation model, like a donut. The chewy 
>> outside parts of the donut are controlled by IntServ. Traffic 
>> leaps across the DiffServ donut-hole only if the heuristic 
>> process thinks (IE, is reasonably sure) that there is 
>adequate bandwidth.
>> 
>>    OK, but I still don't see what that has to do
>>    with proxies. That sounds like something an
>>    edge router would do before it admits diffserv
>>    traffic into the diffserv core. Proxies are even
>>    more clueless because they never see the actual
>>    traffic patterns.
>> 
>> 		Mike
>> Hfæj)bz	b²Ô^>X¬¶ÆÞ-YZnÇ(sm§ÿåSËlmée*¦ìr?¿(tm)¨¥(tm)©ÿ-+-SwèþÈ©
>


_______________________________________________
SIP mailing list
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http://lists.bell-labs.com/mailman/listinfo/sip


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Subject: Re: [SIP] Forcing call setup through home proxy (solution?)
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Adam B. Roach writes:
 > Back in the day, you could manually route e-mail messages through
 > a variety of servers by specifying an email address of the format
 > adam.roach%internal_host@ericsson.com. Upon receiving this,
 > ericsson.com would turn the last % into an @ and continue
 > routing the message. This could be indefinitely chained
 > to produce something like adam.roach%host3%host2%host1@ericsson.com
 > which would route the message through ericsson.com, then host1,
 > then host2, and finally deliver it to host3.

   Would this not lead to the need for a SIP
   version of ORBS?

	      Mike, half serious


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From sip-admin@lists.bell-labs.com  Sun May 21 03:42:48 2000
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "Sinnreich, Henry" <Henry.Sinnreich@WCOM.Com>
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Subject: RE: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task for	cein Adelaide)
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Henry wrote:
> Dean wrote:
> >The real question is to define a mechanism that allows the 
> >health of the network to be assessed in some sort of 
> >meaningful way so that this knowledge may be made available to the UAC.
> 
> The model in <sinnreich-sip-qos-osp> actually deals with this problem.
> Upon failure of the RSVP request, the telephony endpoint can 
> alert the user:
> "QoS cannot be provided, would you like to proceed with the call".

Perhaps, but I'm trying to find a way to get knowledge about the state of the backbone network into an edge device WITHOUT using RSVP.

It just occurred to met that this could also be used as a means of traffic shaping enterprise access to fit within the bounds of a slightly elastic SLA.

Let's say CorpA has an SLA whereby they are allowed to admit up to an average of 1Mb/S of expedited traffic across any 10-minute window, and whatever their port will carry of best-effort.

Here's one possible way:

The access routers feeds class-usage counters periodically (say, 1 minute intervals) to a network management system. This NMS then calculates a sliding average for each traffic class.  From the current average, it calculates the maximal steady-state bandwith allowable over the next ten minutes to stay within the SLA minus a safety margin. The NMS then divides this number by the average bandwith requirement of a phone call at this site, thereby generating a number which is roughly the equivalent of the number of calls that may be admitted RIGHT NOW without causing a violation of the SLA to occur. The NMS posts this data to a proxy, which then decrements this number for each additional call admitted. When a new estimate comes in, it replaces the current working number. If the working number drops to zero (or some low threhold to allow for emergency calls), calls are not admitted unlesss they appropriately marked "urgent".

I suppose one could do a monte-carlo simulation of this and get a reasonable feel for the efficacy.

--
Dean

ÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÒ ùšŠX§‚X¬µ"þX¬¶ÏÛzYÿ•¦ìýÊ&†ÛiÿÿåŠËlý·¥—ùZnÏÜ¢oæj)fjåŠËbú?


From sip-admin@lists.bell-labs.com  Sun May 21 22:41:50 2000
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This technique is the moral equivalent of H.323 GK-based bandwidth management.  The results from that set of experiences leaves the proposition unproven.  To my way of thinking, there are a number of inherent weaknesses in the model (for example, in your scenario, you won't discover anything about the backbone, only something about the access routers).  On the other hand, you don't need anything new in SIP itself to build such a proxy, right?

Cary.

Dean Willis wrote:

> Henry wrote:
> > Dean wrote:
> > >The real question is to define a mechanism that allows the
> > >health of the network to be assessed in some sort of
> > >meaningful way so that this knowledge may be made available to the UAC.
> >
> > The model in <sinnreich-sip-qos-osp> actually deals with this problem.
> > Upon failure of the RSVP request, the telephony endpoint can
> > alert the user:
> > "QoS cannot be provided, would you like to proceed with the call".
>
> Perhaps, but I'm trying to find a way to get knowledge about the state of the backbone network into an edge device WITHOUT using RSVP.
>
> It just occurred to met that this could also be used as a means of traffic shaping enterprise access to fit within the bounds of a slightly elastic SLA.
>
> Let's say CorpA has an SLA whereby they are allowed to admit up to an average of 1Mb/S of expedited traffic across any 10-minute window, and whatever their port will carry of best-effort.
>
> Here's one possible way:
>
> The access routers feeds class-usage counters periodically (say, 1 minute intervals) to a network management system. This NMS then calculates a sliding average for each traffic class.  From the current average, it calculates the maximal steady-state bandwith allowable over the next ten minutes to stay within the SLA minus a safety margin. The NMS then divides this number by the average bandwith requirement of a phone call at this site, thereby generating a number which is roughly the equivalent of the number of calls that may be admitted RIGHT NOW without causing a violation of the SLA to occur. The NMS posts this data to a proxy, which then decrements this number for each additional call admitted. When a new estimate comes in, it replaces the current working number. If the working number drops to zero (or some low threhold to allow for emergency calls), calls are not admitted unlesss they appropriately marked "urgent".
>
> I suppose one could do a monte-carlo simulation of this and get a reasonable feel for the efficacy.
>
> --
> Dean
>
> Hƒæj)bž b²Ôˆ>X¬¶ÆÞ–YZnÇ(šm§ÿåŠËlmée•¦ìr‰¿™¨¥™©ÿ–+-ŠwèþÈ©



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From sip-admin@lists.bell-labs.com  Mon May 22 04:35:46 2000
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From: "Velibor Cakarevic (ECS)" <Velibor.Cakarevic@ecs.ericsson.se>
To: sip@lists.bell-labs.com
Subject: [SIP] Clarification of CSeq
Date: Mon, 22 May 2000 10:10:09 +0200
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In the document draft-ietf-sip-call-flows-00.txt there is an example of a successful SIP to SIP session establishment (chapter 3.1.1).

I wonder if the CSeq should not increase its value by one when a session is terminated with the BYE signal? In the draft example the Invite signal has the value CSeq: 1 INVITE and the bye signal CSeq: 1 BYE. Should it not be CSeq: 2 BYE?  

Assume that a User A sends 'INVITE' to a User B (CSeq: 1 INVITE). User B replies with '486 busy here' (CSeq: 1 INVITE). What should the value of CSeq be in 'ACK' from User A to User B? I think it should be CSeq: 2 ACK.

____________________________________________
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From sip-admin@lists.bell-labs.com  Mon May 22 04:38:13 2000
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From: "Velibor Cakarevic (ECS)" <Velibor.Cakarevic@ecs.ericsson.se>
To: sip@lists.bell-labs.com
Subject: [SIP] Clarification of CSeq
Date: Mon, 22 May 2000 10:10:09 +0200
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In the document draft-ietf-sip-call-flows-00.txt there is an example of a successful SIP to SIP session establishment (chapter 3.1.1).

I wonder if the CSeq should not increase its value by one when a session is terminated with the BYE signal? In the draft example the Invite signal has the value CSeq: 1 INVITE and the bye signal CSeq: 1 BYE. Should it not be CSeq: 2 BYE?  

Assume that a User A sends 'INVITE' to a User B (CSeq: 1 INVITE). User B replies with '486 busy here' (CSeq: 1 INVITE). What should the value of CSeq be in 'ACK' from User A to User B? I think it should be CSeq: 2 ACK.

____________________________________________
Velibor Cakarevic <velibor.cakarevic@ecs.ericsson.se>




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From sip-admin@lists.bell-labs.com  Mon May 22 05:40:45 2000
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From: Gethin Liddell <gethin@ubiquity.net>
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Hi,

can i get confirmation that the mime-version and content-language headers should be present (or not) in the new bis please.

I remember a thread on the validity of the content-language and thought the result was that it should not exist.

It is still present in the new bis (May) along with the mime-version header but neither are mentioned in Appendix F, have a description of their use or are marked as new.

thanx

-- 
Gethin Liddell
Ubiquity Software Corporation

http://www.ubiquity.net
gethin@ubiquity.net


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vc > I wonder if the CSeq should not increase its value by one when a
session is terminated
vc > with the BYE signal? In the draft example the Invite signal has the
value CSeq: 1
vc > INVITE and the bye signal CSeq: 1 BYE. Should it not be CSeq: 2 BYE?


CSeq numbering is maintained separately in each direction.
The sequence nos. from A - B does not relate to the sequence nos. from B-A.

so, A had sent an INVITE - and its seq no is 1
B sends a BYE, and hence its seq no can also be 1 (or whatever else). There
is no relation b/w the two seq nos.
If A sent BYE, then sequence would be 2.



vc >Assume that a User A sends 'INVITE' to a User B (CSeq: 1 INVITE). User
B replies with
vc >'486 busy here' (CSeq: 1 INVITE). What should the value of CSeq be in
'ACK' from User A
vc >to User B? I think it should be CSeq: 2 ACK.


The value should be the same as the INVITE. ie 1, so as to match the
original INVITE-OK transaction to the ACK. (ACK is a new request and its
CSeq method param is changed to ACK, the CSeq helps to associate it to the
OK response)


Regds
Arjun

--
Arjun Roychowdhury @ Hughes Software Systems






"Velibor Cakarevic (ECS)" <Velibor.Cakarevic@ecs.ericsson.se> on 05/22/2000
01:40:09 PM

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

Subject:  [SIP] Clarification of CSeq




In the document draft-ietf-sip-call-flows-00.txt there is an example of a
successful SIP to SIP session establishment (chapter 3.1.1).

I wonder if the CSeq should not increase its value by one when a session is
terminated with the BYE signal? In the draft example the Invite signal has
the value CSeq: 1 INVITE and the bye signal CSeq: 1 BYE. Should it not be
CSeq: 2 BYE?

Assume that a User A sends 'INVITE' to a User B (CSeq: 1 INVITE). User B
replies with '486 busy here' (CSeq: 1 INVITE). What should the value of
CSeq be in 'ACK' from User A to User B? I think it should be CSeq: 2 ACK.

____________________________________________
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From sip-admin@lists.bell-labs.com  Mon May 22 06:54:17 2000
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Date: Mon, 22 May 2000 12:55:36 +0200 (CEST)
From: Lars Berggren <lars@intertex.se>
To: "Schroeder, Tim" <schroeder@tri.sbc.com>
Cc: sip@lists.bell-labs.com
Subject: RE: [SIP] Comments on Session Timer draft
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On Fri, 19 May 2000, Schroeder, Tim wrote:
> At any rate, I'd like the draft to address the cases where any of the
> UAC/UAS/proxy *require* the session timer feature.  I'd consider it OK for
> the session to not get set up if the session timer feature weren't supported
> by all (or enough) of the UAs, but not OK for the session to get set up
> without a timer if one were required anywhere along the path.

I agree. I think any sip device involved in the signalling could be
interested in a way to:
1) Suggest/request/propose that a timer is used.
2) Require a timer.
3) Kind of inform that resources will be released at a certain
time, regardless of whether a session timer is used or not. If no session
timer is used you just cant get 100% service. Like "Services applied in 
this call will time out, send re-INVITE to update the timeout".

I think 1 is fulfilled by the current draft. 2 will be so if any of 
the parties were allowed to add the require header. 3 would be fulfilled
if Session-Expires were allowed to be added by a proxy in both requests
and responses even if the proxy has no way of knowing whether a timer is
supported or not. I cant see the point in disallowing it.

Also, again, I think it is sufficient that one of UAS/UAC is session timer
aware to keep the session alive.

Any comments from the authors?

>  
> Also, the third to the last paragraph in section 4 (beginning "A UAC may use
> the refreshing re-INVITE as a normal SIP re-INVITE ...") makes no sense to
> me.  It talks about sending an *updated* session description and using the
> origin field to indicate the description is *unchanged*.
> 

Yes, that subsection confuses me too. Is it a typo?

/Lars

> Tim Schroeder
> tschroeder@tri.sbc.com
> SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759
> 
>  
> 
> 
> _______________________________________________
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> 

Lars Berggren       <mailto:lars.berggren@intertex.se>
Intertex Data AB    tel: +46-8-6282828
Sundbyberg, Sweden  fax: +46-8-6286414



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Gethin Liddell wrote:
> 
> Hi,
> 
> can i get confirmation that the mime-version and content-language headers should be present (or not) in the new bis please.
> 
> I remember a thread on the validity of the content-language and thought the result was that it should not exist.

I don't think there was such a conclusion. My opinion is that it is an
optional header and thus no harm is done nor is complexity increased by
including it. It may be useful in some limited circumstances. If
information is exported from an HTTP source to a SIP destination, it
seems to make sense to avoid losing information when doing the
exporting. One slightly less contrived use might be within MIME
multiparts, so that a Canadian SIP server could automatically return and
label the English and French versions of the HTML or error response (see
RFC 1766, Section 3.).

MIME-Version again would be optional. I'm not sure if any HTTP servers
actually produce this header. As far as I can tell, it's basically a
no-op, i.e., it doesn't seem to change behavior in the client or server
in any way. I suppose it would matter only if MIME Version 2.0 came
along and we wanted to signal this. (Unlike HTTP, SIP obviously does
make more extensive use of MIME for multipart bodies.)

> 
> It is still present in the new bis (May) along with the mime-version header but neither are mentioned in Appendix F, have a description of their use or are marked as new.
> 
> thanx
> 
> --
> Gethin Liddell
> Ubiquity Software Corporation
> 
> http://www.ubiquity.net
> gethin@ubiquity.net
> 
> _______________________________________________
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-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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From sip-admin@lists.bell-labs.com  Mon May 22 10:11:04 2000
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I'm really not trying to propose the "gatekeeper control" here -- the fundaments of that are well-established. What I'm attempting to do is show how knowledge about the state of the network can be made available to SIP UAs without having to have the UA explicitly sharing (and understanding)that knowledge.

I'm also somewhat muddlingly trying to consider what sort of knowledge might be available at a relatively low cost (both in network and development terms) and how we might apply that knowledge in a variety of SIP environs.

--
Dean

> -----Original Message-----
> From: sip-admin@lists.bell-labs.com
> [mailto:sip-admin@lists.bell-labs.com]On Behalf Of Cary FitzGerald
> Sent: Sunday, May 21, 2000 9:43 PM
> To: Dean Willis
> Cc: Sinnreich, Henry; IETF SIP
> Subject: Re: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task
> for cein Adelaide)
> 
> 
> This technique is the moral equivalent of H.323 GK-based 
> bandwidth management.  The results from that set of experiences 
> leaves the proposition unproven.  To my way of thinking, there 
> are a number of inherent weaknesses in the model (for example, in 
> your scenario, you won't discover anything about the backbone, 
> only something about the access routers).  On the other hand, you 
> don't need anything new in SIP itself to build such a proxy, right?
> 
> Cary.
> 
> Dean Willis wrote:
> 
> > Henry wrote:
> > > Dean wrote:
> > > >The real question is to define a mechanism that allows the
> > > >health of the network to be assessed in some sort of
> > > >meaningful way so that this knowledge may be made available 
> to the UAC.
> > >
> > > The model in <sinnreich-sip-qos-osp> actually deals with this problem.
> > > Upon failure of the RSVP request, the telephony endpoint can
> > > alert the user:
> > > "QoS cannot be provided, would you like to proceed with the call".
> >
> > Perhaps, but I'm trying to find a way to get knowledge about 
> the state of the backbone network into an edge device WITHOUT using RSVP.
> >
> > It just occurred to met that this could also be used as a means 
> of traffic shaping enterprise access to fit within the bounds of 
> a slightly elastic SLA.
> >
> > Let's say CorpA has an SLA whereby they are allowed to admit up 
> to an average of 1Mb/S of expedited traffic across any 10-minute 
> window, and whatever their port will carry of best-effort.
> >
> > Here's one possible way:
> >
> > The access routers feeds class-usage counters periodically 
> (say, 1 minute intervals) to a network management system. This 
> NMS then calculates a sliding average for each traffic class.  
> From the current average, it calculates the maximal steady-state 
> bandwith allowable over the next ten minutes to stay within the 
> SLA minus a safety margin. The NMS then divides this number by 
> the average bandwith requirement of a phone call at this site, 
> thereby generating a number which is roughly the equivalent of 
> the number of calls that may be admitted RIGHT NOW without 
> causing a violation of the SLA to occur. The NMS posts this data 
> to a proxy, which then decrements this number for each additional 
> call admitted. When a new estimate comes in, it replaces the 
> current working number. If the working number drops to zero (or 
> some low threhold to allow for emergency calls), calls are not 
> admitted unlesss they appropriately marked "urgent".
> >
> > I suppose one could do a monte-carlo simulation of this and get 
> a reasonable feel for the efficacy.
> >
> > --
> > Dean
> >
> > Hƒæj)bž b²Ôˆ>X¬¶ÆÞ–YZnÇ(šm§ÿåŠËlmée•¦ìr‰¿™¨¥™©ÿ–+-ŠwèþÈ©
> 
> 
> 
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From: CHAPRON Jean-Emmanuel FTRD/DAC/ISS <jeanemmanuel.chapron@rd.francetelecom.fr>
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Hi,

What's the current status of this draft:
"draft-ietf-sip-info-method-03.txt"? 

I read there was a Last Call on version 2; and on version 3? Is it soon
approved by the IESG?
Actually, I think I missed something.


Thanks for your help

               ```
              (o o)
----------oOO--(_)-OOo--------------
Jean-Emmanuel CHAPRON
FRANCE TELECOM R&D
DAC/ARS
38-40 rue du général Leclerc
92794 Issy-les-Moulineaux Cedex 9 - France
Tel: +33 1 45 29 64 50
-------------------------------------



> -----Message d'origine-----
> De: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> Date: lundi 10 avril 2000 13:35
> Cc: sip@lists.research.bell-labs.com
> Objet: [SIP] I-D ACTION:draft-ietf-sip-info-method-03.txt
> 
> 
> A New Internet-Draft is available from the on-line 
> Internet-Drafts directories.
> This draft is a work item of the Session Initiation Protocol 
> Working Group of the IETF.
> 
> 	Title		: The SIP INFO Method
> 	Author(s)	: S. Donovan
> 	Filename	: draft-ietf-sip-info-method-03.txt
> 	Pages		: 10
> 	Date		: 06-Apr-00
> 	
> This document proposes an extension to the Session Initiation
> Protocol (SIP).  This extension adds the INFO method to the SIP
> protocol.  The intent of the INFO method is to allow for the carrying
> of session related control information that is generated during a
> session.  One example of such session control information is ISUP and
> ISDN signaling messages used to control telephony call services.
> This and other example uses of the INFO method may be standardized in
> the future.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-info-method-03.txt
> 
> Internet-Drafts are also available by anonymous FTP. Login 
> with the username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
> 	"get draft-ietf-sip-info-method-03.txt".
> 
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html 
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 
> 
> Internet-Drafts can also be obtained by e-mail.
> 
> Send a message to:
> 	mailserv@ietf.org.
> In the body type:
> 	"FILE /internet-drafts/draft-ietf-sip-info-method-03.txt".
> 	
> NOTE:	The mail server at ietf.org can return the document in
> 	MIME-encoded form by using the "mpack" utility.  To use this
> 	feature, insert the command "ENCODING mime" before the "FILE"
> 	command.  To decode the response(s), you will need "munpack" or
> 	a MIME-compliant mail reader.  Different MIME-compliant 
> mail readers
> 	exhibit different behavior, especially when dealing with
> 	"multipart" MIME messages (i.e. documents which have been split
> 	up into multiple messages), so check your local documentation on
> 	how to manipulate these messages.
> 		
> 		
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
> 


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From: James Kempf <James.Kempf@Eng.Sun.COM>
Reply-To: James Kempf <James.Kempf@Eng.Sun.COM>
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
To: giuseppe.ricagni@italtel.it, mat@cisco.com
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> >    * When the callee is roaming, either you use a standardized database 
transport protocol or you
> >      delegate the callee home domain proxy to perform the query, but you 
have a RTT towards the callee
> >      home domain. 
>
>   Which may effectively be zero.
>
>> Your solution of having synchronized copies of such database all over the 
world is
> >      another interesting enhancement, maybe expensive (pretty sure 
off-the-shelf solutions are already
> >      out there)
>
>   It doesn't sound much more interesting than a
>   localize LDAP cache on the client proxy. There
>   are other ways to do this as well. And it doesn't
>   imply synchronized; all that is necessary are caches
>   with TTL's.
>
> >         o the point I was making was that for all subsequent mid-call 
messages, just like the INVITEs
> >           you have to send for handovers, you should shortcut and avoid the 
round trip to the home
> >           net.
>
>   I disagree. Most traffic is between the two UA's.
>   And it absolutely does not follow that proxies 
>   need to have any part in handoffs. In fact, I
>   do not believe that proxies have any part in
>   handoffs when the UA is in the mobile unit. 
>
> >    * When the caller is roaming, under certain circumstances (related to the 
type of value added
> >      services required & the set of services supported by the visited 
network &  the possibility to
> >      transfer a user profile--see my previous messages)
>
>   I don't know what services the roaming network
>   wants to provide -- other than trying to shoehorn
>   themselves into the call. Even if, there is nothing
>   to prevent me from obtaining those service even if
>   it's routed through my home proxy first.
>


What about having a standardized XML-based user profile description?
That way, the guy that wants to set up a telephony provider in
his garage can have the same access to a description of his
services as anybody else.

Not that this has anything to do with SIP...

		jak



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From sip-admin@lists.bell-labs.com  Mon May 22 12:42:21 2000
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Date: Mon, 22 May 2000 12:24:47 -0400
From: Hyong Sop Shim <hyongsop@research.telcordia.com>
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: [SIP] Multimedia Conferencing using SIP
References: <410B13D70EB2D211827E00E0291E9FF7B2BCF4@ORT-MAIL> <39238277.963763CC@dynamicsoft.com> <39258FFF.6D51A10A@research.telcordia.com> <3925A7C3.D1422138@dynamicsoft.com>
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In the case of the conferences based on "dial-up" bridges, the current model is that
the invited user contacts the dial-up bridge as a result of receving an INVITE message with an Also header sent by an existing conference participant.  Instead, why not have
the dial-up bridge directly invite the invited user?  That is, why not have the existing
conference member send the same INVITE with an Also header to the dial-up bridge
and have the bridge "relay" the INVITE to the user specified in the Also header?
This approach would eliminate the need for the dial-up bridge having to subsume call
legs and provide more a means for allerting the invited user of the current membership
of the conference than RCTP.

Thanks,

--Hyong

Jonathan Rosenberg wrote:

> Hyong Sop Shim wrote:
> >
> > Jonathan Rosenberg wrote:
> >
> > > No extensions needed for central conference servers. It works just like
> > > it does in the PSTN - you dial a number which corresponds to a port on a
> > > bridge. In the case of SIP, you would INVITE to a URL corresponding to a
> > > port on the conference server, like sip:conference33287372@mcu.com. The
> > > result appears as a point to point SIP call as far as  your UA is
> > > concerned, but the server is mixing media and sending the mixed media to
> > > your UA. You will learn the identity of the other participants through
> > > RTCP CSRC.
> > >
> > > How do you know the right URL to send an INVITE to? Well, how do you
> > > know what number to dial for a conference in PSTN? You're told out of
> > > band. URLs can be included in emails, web pages, whatever.
> > >
> >
> > In "SIP Call Control Services," the underlying model for establishing a conference seems to be that it is the responsibility of the invited user to send INVITE requests to the existing conference participants.
>
> That is one model, but not the only one SIP provides. SIP has four ways
> to do multiparty conferencing:
>
> 1. dialup conference bridge: call the bridge just like you call a
> person. Conference is identified by request URI. Works with rfc2543 - no
> extensions needed.
> 2. distributed multiparty conferencing - no server. Fully distributed.
> This is what is described in the now-expired draft, and is work in
> progress.
> 3. multicast conferences - you can run your conference on multicast.
> Simply INVITE the person to join the multicast session. Works with
> baseline SIP. In fact, this was the initial purpose of SIP. In this
> case, there is not a full mesh of SIP signaling.
> 4. 3-way with local MC/MP function: A and B are talking. A wishes to add
> C to the call. A can simply call C, but also act as a mixer, so that the
> media it sends to B contains the A+C media, and the media to C is the
> A+B media. RTCP CSRC indicate who is in the conference. This also works
> with baseline SIP. It imposes additional burden on the UA, but otherwise
> provides this standard feature in a simple way.
>
> There is no preferred way. It depends on the particular application.
>
> -Jonathan R.
>
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com



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To: James Kempf <James.Kempf@Eng.Sun.COM>
Cc: giuseppe.ricagni@italtel.it, mat@cisco.com, sip@lists.bell-labs.com
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James Kempf writes:
 > >   I don't know what services the roaming network
 > >   wants to provide -- other than trying to shoehorn
 > >   themselves into the call. Even if, there is nothing
 > >   to prevent me from obtaining those service even if
 > >   it's routed through my home proxy first.
 > >
 > 
 > 
 > What about having a standardized XML-based user profile description?
 > That way, the guy that wants to set up a telephony provider in
 > his garage can have the same access to a description of his
 > services as anybody else.

   I think we have arrived at an impasse since I
   view this as a bug, not a feature. A standardized
   scheme always locks you to the least common
   denominator when you want to remote the
   features. That sounds like a shame to me; if I
   run it through my home proxy, I can have as
   many whacko features as I want to cook up
   without having to convince the world that
   they're a Good Idea. That way, the market
   decides rather than a bunch of standards
   weenies (present company excluded, yadda
   yadda).

		Mike


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From sip-admin@lists.bell-labs.com  Mon May 22 16:26:47 2000
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Date: Mon, 22 May 2000 20:24:55 +0000
From: "Sinnreich, Henry" <Henry.Sinnreich@WCOM.Com>
Subject: RE: [SIP] Minutes of SIP Security task force in Adelaide
To: "'Michael Thomas'" <mat@cisco.com>
Cc: giuseppe.ricagni@italtel.it, sip@lists.bell-labs.com,
        James Kempf <James.Kempf@Eng.Sun.COM>
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>That way, the market
>   decides rather than a bunch of standards
>   weenies

Wow Michael! The Internet is defined entirely by its standard protocols,
developed with great care by 'weenies', and not by 'market decides'
managers. We know a lot of protocols developed by 'market decides' managers
that are now history.

>>   view this as a bug, not a feature.
Actually, the portability of the service may seem very attractive to
customers, and is one of the SIP potential applications. I could upload my
preferences to the convenient service when on travel for example, to reduce
some costs. Also, I could change these preferences when on travel, so as to
adjust them to my changed needs. Enlightened service providers that allow
such conveniences may be preferred by customers.

My two cents, Henry

>-----Original Message-----
>From: Michael Thomas [mailto:mat@cisco.com]
>Sent: Monday, May 22, 2000 2:55 PM
>To: James Kempf
>Cc: giuseppe.ricagni@italtel.it; mat@cisco.com; sip@lists.bell-labs.com
>Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
>
>
>James Kempf writes:
> > >   I don't know what services the roaming network
> > >   wants to provide -- other than trying to shoehorn
> > >   themselves into the call. Even if, there is nothing
> > >   to prevent me from obtaining those service even if
> > >   it's routed through my home proxy first.
> > >
> > 
> > 
> > What about having a standardized XML-based user profile description?
> > That way, the guy that wants to set up a telephony provider in
> > his garage can have the same access to a description of his
> > services as anybody else.
>
>   I think we have arrived at an impasse since I
>   view this as a bug, not a feature. A standardized
>   scheme always locks you to the least common
>   denominator when you want to remote the
>   features. That sounds like a shame to me; if I
>   run it through my home proxy, I can have as
>   many whacko features as I want to cook up
>   without having to convince the world that
>   they're a Good Idea. That way, the market
>   decides rather than a bunch of standards
>   weenies (present company excluded, yadda
>   yadda).
>
>		Mike
>
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip
>


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From sip-admin@lists.bell-labs.com  Mon May 22 16:34:27 2000
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From: James Kempf <James.Kempf@eng.sun.com>
Reply-To: James Kempf <James.Kempf@eng.sun.com>
Subject: RE: [SIP] Minutes of SIP Security task force in Adelaide
To: mat@cisco.com, Henry.Sinnreich@WCOM.Com
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>
>>That way, the market
>>   decides rather than a bunch of standards
>>   weenies
>
>Wow Michael! The Internet is defined entirely by its standard protocols,
>developed with great care by 'weenies', and not by 'market decides'
>managers. We know a lot of protocols developed by 'market decides' managers
>that are now history.
>
>>>   view this as a bug, not a feature.
>Actually, the portability of the service may seem very attractive to
>customers, and is one of the SIP potential applications. I could upload my
>preferences to the convenient service when on travel for example, to reduce
>some costs. Also, I could change these preferences when on travel, so as to
>adjust them to my changed needs. Enlightened service providers that allow
>such conveniences may be preferred by customers.
>

Right. The other thing a standardized user profile format would let me do is 
edit services on any  Web browser rather than have to call up the phone company 
and wait for them change my service profile. In New World Telephony,
this empowers the user so that they effectively control their services.
The phone company benefits in that their administrative costs are reduced.
And if the XML DTD for the user profile is made extensible, phone
service providers can add customized features, just like Web pages
have different features.

		jak


>My two cents, Henry
>
>>-----Original Message-----
>>From: Michael Thomas [mailto:mat@cisco.com]
>>Sent: Monday, May 22, 2000 2:55 PM
>>To: James Kempf
>>Cc: giuseppe.ricagni@italtel.it; mat@cisco.com; sip@lists.bell-labs.com
>>Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
>>
>>
>>James Kempf writes:
>> > >   I don't know what services the roaming network
>> > >   wants to provide -- other than trying to shoehorn
>> > >   themselves into the call. Even if, there is nothing
>> > >   to prevent me from obtaining those service even if
>> > >   it's routed through my home proxy first.
>> > >
>> > 
>> > 
>> > What about having a standardized XML-based user profile description?
>> > That way, the guy that wants to set up a telephony provider in
>> > his garage can have the same access to a description of his
>> > services as anybody else.
>>
>>   I think we have arrived at an impasse since I
>>   view this as a bug, not a feature. A standardized
>>   scheme always locks you to the least common
>>   denominator when you want to remote the
>>   features. That sounds like a shame to me; if I
>>   run it through my home proxy, I can have as
>>   many whacko features as I want to cook up
>>   without having to convince the world that
>>   they're a Good Idea. That way, the market
>>   decides rather than a bunch of standards
>>   weenies (present company excluded, yadda
>>   yadda).
>>
>>		Mike
>>
>>
>>_______________________________________________
>>SIP mailing list
>>SIP@lists.bell-labs.com
>>http://lists.bell-labs.com/mailman/listinfo/sip
>>
>
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip



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From sip-admin@lists.bell-labs.com  Mon May 22 16:35:24 2000
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Date: Mon, 22 May 2000 16:36:19 -0400
To: Michael Thomas <mat@cisco.com>, James Kempf <James.Kempf@Eng.Sun.COM>
From: "Michael A. Ramalho" <mramalho@cisco.com>
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
Cc: giuseppe.ricagni@italtel.it, mat@cisco.com, sip@lists.bell-labs.com
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James/Mike,

I, for one, am grateful that you two have realized that
this is the crux of the debate between you. Can we go
on now?

What Mike Thomas says is correct .... the standardized
scheme will lock you into some core or least common
denominator services.

Because the relative scarcity of really neat features in
the wireless world ... some architects of tremendously
successful wireless deployments (e.g., GSM) believe
that their success was in the standardization of profiles.

I think they are/were correct from the perspective of
billing (and the corresponding roaming agreements) and
some traditional "old world" PSTN services. [When once
in a vacuum, any particulate matter looks interesting.]

However, it is *very* clear to me that in the "new world"
of web-based or web-like services (and services definition)
that standardization of anything but the most rudimentary
subscriber information or core services will not make it in
"new world" deployments.

Don't fence me in,

Michael Ramalho

At 12:55 PM 5/22/00 -0700, Michael Thomas wrote:
>James Kempf writes:
> > >   I don't know what services the roaming network
> > >   wants to provide -- other than trying to shoehorn
> > >   themselves into the call. Even if, there is nothing
> > >   to prevent me from obtaining those service even if
> > >   it's routed through my home proxy first.
> > >
> > 
> > 
> > What about having a standardized XML-based user profile description?
> > That way, the guy that wants to set up a telephony provider in
> > his garage can have the same access to a description of his
> > services as anybody else.
>
>   I think we have arrived at an impasse since I
>   view this as a bug, not a feature. A standardized
>   scheme always locks you to the least common
>   denominator when you want to remote the
>   features. That sounds like a shame to me; if I
>   run it through my home proxy, I can have as
>   many whacko features as I want to cook up
>   without having to convince the world that
>   they're a Good Idea. That way, the market
>   decides rather than a bunch of standards
>   weenies (present company excluded, yadda
>   yadda).
>
>		Mike
>
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip
>


Michael A. Ramalho, Ph.D.
Cisco Systems
1802 Rue de la Port
Wall Township, NJ 07719-3784
Office email: mramalho@cisco.com
Personal email: mar42@cornell.edu
Office: +1.732.449.5762
Cell: +1.732.809.0188
Pager: +1.800.365.4578


_______________________________________________
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From sip-admin@lists.bell-labs.com  Mon May 22 16:39:09 2000
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Date: Mon, 22 May 2000 13:41:21 -0700 (PDT)
From: James Kempf <James.Kempf@eng.sun.com>
Reply-To: James Kempf <James.Kempf@eng.sun.com>
Subject: Re: [SIP] Minutes of SIP Security task force in Adelaide
To: mat@cisco.com, James.Kempf@eng.sun.com, mramalho@cisco.com
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>However, it is *very* clear to me that in the "new world"
>of web-based or web-like services (and services definition)
>that standardization of anything but the most rudimentary
>subscriber information or core services will not make it in
>"new world" deployments.
>
>Don't fence me in,
>

Don't see why this should be any more the case for a telephony
DTD than for any other standardized XML document. Naturally,
there needs to be some provision in the format for extensibility,
and I suspect that Old World Telephony GSM standardization didn't
see the need for that. 

		jak


>Michael Ramalho
>
>At 12:55 PM 5/22/00 -0700, Michael Thomas wrote:
>>James Kempf writes:
>> > >   I don't know what services the roaming network
>> > >   wants to provide -- other than trying to shoehorn
>> > >   themselves into the call. Even if, there is nothing
>> > >   to prevent me from obtaining those service even if
>> > >   it's routed through my home proxy first.
>> > >
>> > 
>> > 
>> > What about having a standardized XML-based user profile description?
>> > That way, the guy that wants to set up a telephony provider in
>> > his garage can have the same access to a description of his
>> > services as anybody else.
>>
>>   I think we have arrived at an impasse since I
>>   view this as a bug, not a feature. A standardized
>>   scheme always locks you to the least common
>>   denominator when you want to remote the
>>   features. That sounds like a shame to me; if I
>>   run it through my home proxy, I can have as
>>   many whacko features as I want to cook up
>>   without having to convince the world that
>>   they're a Good Idea. That way, the market
>>   decides rather than a bunch of standards
>>   weenies (present company excluded, yadda
>>   yadda).
>>
>>		Mike
>>
>>
>>_______________________________________________
>>SIP mailing list
>>SIP@lists.bell-labs.com
>>http://lists.bell-labs.com/mailman/listinfo/sip
>>
>
>
>Michael A. Ramalho, Ph.D.
>Cisco Systems
>1802 Rue de la Port
>Wall Township, NJ 07719-3784
>Office email: mramalho@cisco.com
>Personal email: mar42@cornell.edu
>Office: +1.732.449.5762
>Cell: +1.732.809.0188
>Pager: +1.800.365.4578



_______________________________________________
SIP mailing list
SIP@lists.bell-labs.com
http://lists.bell-labs.com/mailman/listinfo/sip


From sip-admin@lists.bell-labs.com  Mon May 22 19:59:44 2000
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From: "Lyndon Ong" <long@nortelnetworks.com>
Subject: RE: [SIP] SIP ISUP MIME type
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Folks,

Apart from Kevin's comment on explicitly specifying the starting boundary
for an encapsulated ISUP message, are there any other comments on
draft-ietf-sip-isup-mime-01.txt?  

Cheers,

L. Ong




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James Kempf wrote:
> 
> >
> >>That way, the market
> >>   decides rather than a bunch of standards
> >>   weenies
> >
> >Wow Michael! The Internet is defined entirely by its standard protocols,
> >developed with great care by 'weenies', and not by 'market decides'
> >managers. We know a lot of protocols developed by 'market decides' managers
> >that are now history.
> >
> >>>   view this as a bug, not a feature.
> >Actually, the portability of the service may seem very attractive to
> >customers, and is one of the SIP potential applications. I could upload my
> >preferences to the convenient service when on travel for example, to reduce
> >some costs. Also, I could change these preferences when on travel, so as to
> >adjust them to my changed needs. Enlightened service providers that allow
> >such conveniences may be preferred by customers.
> >
> 
> Right. The other thing a standardized user profile format would let me do is
> edit services on any  Web browser rather than have to call up the phone company
> and wait for them change my service profile. In New World Telephony,
> this empowers the user so that they effectively control their services.
> The phone company benefits in that their administrative costs are reduced.
> And if the XML DTD for the user profile is made extensible, phone
> service providers can add customized features, just like Web pages
> have different features.

There is certainly room for both.

CPL is aimed at exactly this kind of service mobility. Its a well
specified API, safe to execute in third party application servers, and
enables some nice services. However, its very limited. It cannot do many
of the exciting new services I think we will see down the road. These
services require more complex APIs and custom services that probably
won't be portable.

So, users can go to their home servers when they want these more
advanced services, and they can use anyone to provide these vanilla,
portable ones.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Subject: Re: [SIP] Multimedia Conferencing using SIP
References: <410B13D70EB2D211827E00E0291E9FF7B2BCF4@ORT-MAIL> <39238277.963763CC@dynamicsoft.com> <39258FFF.6D51A10A@research.telcordia.com> <3925A7C3.D1422138@dynamicsoft.com> <39295F4F.4227A45@research.telcordia.com>
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Hyong Sop Shim wrote:
> 
> In the case of the conferences based on "dial-up" bridges, the current model is that
> the invited user contacts the dial-up bridge as a result of receving an INVITE message with an Also header sent by an existing conference participant. 

Not neccesarily. The most common model for dial-up bridges is that you
simply know the conference ID ahead of time, and call in. Just like in
the PSTN today. This doesn't require any SIP extensions, Also or
anything else.

 Instead, why not have
> the dial-up bridge directly invite the invited user?  That is, why not have the existing
> conference member send the same INVITE with an Also header to the dial-up bridge
> and have the bridge "relay" the INVITE to the user specified in the Also header?
> This approach would eliminate the need for the dial-up bridge having to subsume call
> legs and provide more a means for allerting the invited user of the current membership
> of the conference than RCTP.

Either works. One is more of a consultation feature (call the user, then
transfer to the bridge), the other is a third party mechanism (call the
bridge, transfer to the user).

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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CHAPRON Jean-Emmanuel FTRD/DAC/ISS wrote:
> 
> Hi,
> 
> What's the current status of this draft:
> "draft-ietf-sip-info-method-03.txt"?

I was told informally by IESG that it was approved, but I don't think I
ever saw the formal announcement yet.

> 
> I read there was a Last Call on version 2; and on version 3? Is it soon
> approved by the IESG?

THe last call was issued on v2, comments from IESG (very minor ones)
were incorporated into v3, which was submitted and approved.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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"Schroeder, Tim" wrote:
> 
> The session timer draft (draft-ietf-sip-session-timer-01.txt) as written is
> asymmetric, in that it seems to assume that the UAC is more likely to
> impolitely abandon the session, and that the proxies and UAS need some way
> to recover from that.
> 
> I'd like to see the reverse case addressed as well: where the UAC (or a
> proxy) is distrustful of the UAS, and needs some way to clean up if the UAS
> impolitely abandons the session.
> 
> I suppose the keep-alive re-INVITES can still be sent by the UAC, and it
> will be the lack of a response (rather than the lack of an incoming message)
> which informs the UAC and the proxies that the UAS has dropped off.

Exactly. Failure of either UAC or UAS is detected with equal ease this
way.

> 
> But in regard to the Open Issues list, I think the answer to the second
> issue (should we allow the UAS to insert a Require: timer header?)

I think you mean UAC.

> should be
> YES.  Just like the UAS can reject a call that won't be set up with a
> session timer, the UAC should be able to avoid having a call set up without
> a session timer.  It would do this by using the "Require: timer" header
> rather than the weaker "Supported: timer" header.  If the UAS cannot handle
> the feature, the call would be rejected, rather than set up without the
> (demanded) timer feature.

I recall that there was consensus to do this. Its easy enough anyway;
you could do it even if the draft didn't explicitly specify it.


> 
> And for the third Open Issue on the list, a proxy should also be able to add
> the "Require: timer" header if the UAC merely "supports" but does not
> "require" the feature.  Just like the proxy can tighten up the timing
> requirements by lowering the expiration time, it should be able to tighten
> up the requirements by making the feature "required" rather than merely
> "supported".  Then, if the UAS did not implement that feature, the request
> would be denied.  [This might take some more thought; it would seem odd to
> the UAC for its request to end up denied for lack of a feature it did not
> require.]

This was discussed during the meeting and on the mailing list prior to
the meeting. The consensus was that we would allow this. A UAC
implementing session timer (i.e., supported) would need to be prepared
to receive a rejection response with code 420 even though it hadn't
inserted a Require header.

> 
> Alternatively, maybe there's no support needed in the UAS at all: the
> re-INVITES are all sent from the UAC, and the UAS needn't know anything
> about the feature to process them.  This is similar to what Lars Berggren  <
> <mailto:lars.berggren@intertex.se> mailto:lars.berggren@intertex.se>
> proposes:

I recall thinking about this possibility, but decided it was tricky. The
other thing the UAS must do that is not mentioend in Lars' mail is
reflect the Session-Expires header. If a UAS doesn't support session
timer, the response has no Session-Expires header, with the result is
that the proxies won't know the actual expiration time. 

> 
> >>
> For the Session Timer to be applied for a call leg, the specifications in
> the draft requires that both the UAC and UAS understand the extension
> (See example 8.5). However, since the keep-alive mechanism is achieved by
> sending 'standard' re-INVITEs, it is sufficient (and useful) that at least
> one of the parties is aware of the Session Timer to accomplish a
> keep-alive mechanism. The usefulness of such an approach is for example
> when a UAC supports 'timer', a stateful proxy server, SPS, wants to apply
> a Session Timer to the call and the UAS doesn't understand the extension.
> 
> Here, the call will be setup without Session Timer even though the SPS
> wanted one and the UAC could do the necessary re-inviting. The UAS doesn't
> need to know about Session Timers in order to process re-INVITEs properly.
> The only thing that forces that the UAS have to be aware of the Session
> Timer is the statement in the draft that the UAS SHOULD send BYE if no
> re-INVITE has arrived when the session expires. Well, the UAC could do the
> sending of the BYE if it does not intend to update the timer.
> 
> So, to solve this problem, I would like to propose that the SPS could add
> a Session-Expires header in the response if the SPS knows the UAC supports
> 'timer' and the response does not contain a Session-Expires header.
> <<
> 
> At any rate, I'd like the draft to address the cases where any of the
> UAC/UAS/proxy *require* the session timer feature.  I'd consider it OK for
> the session to not get set up if the session timer feature weren't supported
> by all (or enough) of the UAs, but not OK for the session to get set up
> without a timer if one were required anywhere along the path.
> 
> Also, the third to the last paragraph in section 4 (beginning "A UAC may use
> the refreshing re-INVITE as a normal SIP re-INVITE ...") makes no sense to
> me.  It talks about sending an *updated* session description and using the
> origin field to indicate the description is *unchanged*.

Thats an error. Thanks for pointing it out.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Lars Berggren wrote:
> 
> On Fri, 19 May 2000, Schroeder, Tim wrote:
> > At any rate, I'd like the draft to address the cases where any of the
> > UAC/UAS/proxy *require* the session timer feature.  I'd consider it OK for
> > the session to not get set up if the session timer feature weren't supported
> > by all (or enough) of the UAs, but not OK for the session to get set up
> > without a timer if one were required anywhere along the path.
> 
> I agree. I think any sip device involved in the signalling could be
> interested in a way to:
> 1) Suggest/request/propose that a timer is used.
> 2) Require a timer.
> 3) Kind of inform that resources will be released at a certain
> time, regardless of whether a session timer is used or not. If no session
> timer is used you just cant get 100% service. Like "Services applied in
> this call will time out, send re-INVITE to update the timeout".
> 
> I think 1 is fulfilled by the current draft. 2 will be so if any of
> the parties were allowed to add the require header. 3 would be fulfilled
> if Session-Expires were allowed to be added by a proxy in both requests
> and responses even if the proxy has no way of knowing whether a timer is
> supported or not. I cant see the point in disallowing it.

But the proxy does have a way of knowing - the Supported header in the
request, coupled with reflection of Session-Expires in the response. I
see no point in inserting the header if the proxy knows its not
understood.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From: "Joshua Moloney" <jmoloney@nortelnetworks.com>
To: "Lyndon Ong" <long@nortelnetworks.com>
Cc: sip@lists.bell-labs.com
Subject: RE: [SIP] SIP ISUP MIME type
Date: Tue, 23 May 2000 08:29:31 +0100
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This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

------_=_NextPart_001_01BFC488.AA916A5A
Content-Type: text/plain

Only a minor comment suggesting that the Content-Transfer-Encoding header be
removed
(see thread "[SIP] Comments on draft-ietf-sip-isup-mime-01", dated 10th
May).

Regards,

Josh Moloney
Nortel Networks

> -----Original Message-----
> From:	Lyndon Ong [SMTP:long@nortelnetworks.com]
> Sent:	23 May 2000 01:00
> To:	sip@lists.bell-labs.com
> Subject:	RE: [SIP] SIP ISUP MIME type
> 
> Folks,
> 
> Apart from Kevin's comment on explicitly specifying the starting boundary
> for an encapsulated ISUP message, are there any other comments on
> draft-ietf-sip-isup-mime-01.txt?  
> 
> Cheers,
> 
> L. Ong
> 
> 
> 

------_=_NextPart_001_01BFC488.AA916A5A
Content-Type: text/html
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2651.65">
<TITLE>RE: [SIP] SIP ISUP MIME type</TITLE>
</HEAD>
<BODY>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Only a minor comment =
suggesting that the Content-Transfer-Encoding header be removed</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">(see thread =
&quot;[SIP] Comments on draft-ietf-sip-isup-mime-01&quot;, dated 10th =
May).</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Regards,</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Josh Moloney</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Nortel =
Networks</FONT>
</P>
<UL>
<P><FONT SIZE=3D1 FACE=3D"Arial">-----Original Message-----</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">From:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Lyndon Ong =
[SMTP:long@nortelnetworks.com]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">23 May 2000 01:00</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">sip@lists.bell-labs.com</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT>=
</B> <FONT SIZE=3D1 FACE=3D"Arial">RE: [SIP] SIP ISUP MIME type</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Folks,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Apart from Kevin's comment on =
explicitly specifying the starting boundary</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">for an encapsulated ISUP message, are =
there any other comments on</FONT>
<BR><FONT SIZE=3D2 =
FACE=3D"Arial">draft-ietf-sip-isup-mime-01.txt?&nbsp; </FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Cheers,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">L. Ong</FONT>
</P>
<BR>
<BR>
</UL>
</BODY>
</HTML>
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From sip-admin@lists.bell-labs.com  Tue May 23 04:09:32 2000
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Date: Tue, 23 May 2000 10:11:01 +0200 (CEST)
From: Lars Berggren <lars@intertex.se>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: "Schroeder, Tim" <schroeder@tri.sbc.com>, sip@lists.bell-labs.com
Subject: Re: [SIP] Comments on Session Timer draft
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On Tue, 23 May 2000, Jonathan Rosenberg wrote:
> Lars Berggren wrote:
> > I think 1 is fulfilled by the current draft. 2 will be so if any of
> > the parties were allowed to add the require header. 3 would be fulfilled
> > if Session-Expires were allowed to be added by a proxy in both requests
> > and responses even if the proxy has no way of knowing whether a timer is
> > supported or not. I cant see the point in disallowing it.
> 
> But the proxy does have a way of knowing - the Supported header in the
> request, coupled with reflection of Session-Expires in the response. I
> see no point in inserting the header if the proxy knows its not
> understood.
> 

OK, as the current draft is written, yes. But if it was changed according
to my proposal which started this thread, no.

Assume the draft was changed to state that either the UAC or the UAS
handles the sending of re-INVITE and that only one of the parties need to
be session timer aware. That is, a timer can be used even if the UAC
does not support timer. The party which sends the re-INVITEs is determined
by the content of the Supported header, if present. If the UAC claims
support for 'timer' it shall do the re-inviting. Otherwise the UAS does
it, if it supports it.

Now, with this assumptions, when a request without support for
'timer' arrives at a proxy it could add a Session-Expires header and hope
for the UAS to support it, but it has no way of knowing for sure.

Also, when a response arriving at a proxy without Session-Expires, but the
proxy knows the request claimed support for 'timer', it would be very
useful if the proxy was allowed to insert a Session-Expires header in the
response.

Instead of just allowing the proxy to add the header in these special 
cases, it could be allowed to be added in any situation since it does not
break anything.

/Lars

> -Jonathan R.
> -- 
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

Lars Berggren       <mailto:lars.berggren@intertex.se>
Intertex Data AB    tel: +46-8-6282828
Sundbyberg, Sweden  fax: +46-8-6286414



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Date: Tue, 23 May 2000 10:28:20 +0200 (CEST)
From: Lars Berggren <lars@intertex.se>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: "Schroeder, Tim" <schroeder@tri.sbc.com>, sip@lists.bell-labs.com
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On Tue, 23 May 2000, Jonathan Rosenberg wrote:

> 
> > 
> > Alternatively, maybe there's no support needed in the UAS at all: the
> > re-INVITES are all sent from the UAC, and the UAS needn't know anything
> > about the feature to process them.  This is similar to what Lars Berggren  <
> > <mailto:lars.berggren@intertex.se> mailto:lars.berggren@intertex.se>
> > proposes:
> 
> I recall thinking about this possibility, but decided it was tricky. The
> other thing the UAS must do that is not mentioend in Lars' mail is
> reflect the Session-Expires header. If a UAS doesn't support session
> timer, the response has no Session-Expires header, with the result is
> that the proxies won't know the actual expiration time. 
> 

Proxies that insist on knowing the actual expiration time can include a
'Require: timer' in the request. 

For other proxies I suppose the Session-Expires could be added and 
eventually decremented in the response-path as well. The proxies will know
that the expiration time is not greater than the one they saw/added. And
if it is needed, the UAC could reflect the actual expiration time in the
ACK. It is not tricky.

/Lars


Lars Berggren       <mailto:lars.berggren@intertex.se>
Intertex Data AB    tel: +46-8-6282828
Sundbyberg, Sweden  fax: +46-8-6286414



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Date: Tue, 23 May 2000 09:44:04 +0100
To: James Kempf <James.Kempf@eng.sun.com>
From: Graham Klyne <GK@Dial.pipex.com>
Cc: mat@cisco.com, mramalho@cisco.com, giuseppe.ricagni@italtel.it,
        sip@lists.bell-labs.com
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At 01:41 PM 5/22/00 -0700, James Kempf wrote:

> >However, it is *very* clear to me that in the "new world"
> >of web-based or web-like services (and services definition)
> >that standardization of anything but the most rudimentary
> >subscriber information or core services will not make it in
> >"new world" deployments.
> >
> >Don't fence me in,
> >
>
>Don't see why this should be any more the case for a telephony
>DTD than for any other standardized XML document. Naturally,
>there needs to be some provision in the format for extensibility,

[...]

There has been much talk over recent months of the Internet's end-to-end 
architecture, and the way that it allows innovation by any connected party 
rather than just those that control the core.  (I think SIP is just one 
fruit of this realization.)

I think there is a new front emerging over which the tensions between 
centralized and decentralized control will occur:  content formats.  How 
many organizations are stuck with a certain vendor's software because their 
documents are in the format of that software?  Through XML, XML namespaces, 
RDF and other initiatives, W3C are working on content formats that offer 
the potential for an end-to-end content architecture, much as IP has 
provided an end-to-end communication architecture.  The key is something 
that Tim Berners-Lee has called "language mixing".  (see 
http://www.w3.org/DesignIssues/Evolution.html)

These ideas are being developed in the context of WWW, but I belive are 
applicable to other Interbnet applications.

In summary:  using the right tools, a definition of core information should 
not be a fence, but a springboard.  (In this, I am broadly agreeing with 
James.)

#g


------------
Graham Klyne
(GK@ACM.ORG)



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From sip-admin@lists.bell-labs.com  Tue May 23 10:49:10 2000
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Subject: [SIP] Forking Proxy Behaviour w.r.t. Multiple 401 & 407 responses
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Hi,

IMHO various sections within the new version of the bis draft should
be updated to reflect the changes made to the paragraph dealing
with 4xx,5xx response in section 12.4, pg 84.

The modification deals with how a forking proxy should handle multiple
authentication responses.

This modification means that SIP authentication moves further away
from the process defined in rfc2617, HTTP Authentication: Basic and
Digest Access Authentication.

Perhaps section 14.1 in the bis draft should include a list of
general differences w.r.t. SIP Authentication.

The topics I'd like to see mentioned are:

1) Now possible for user agent to receive challenges from more than
   1 realm in a single response. This contradicts last paragraph of
   section 3.6 in rfc2617.

2) Possible to receive Proxy-Authenticate header in a 401 response.

   That said, it's not explicitly stated in rfc2617 that a 401 response
   can't contain a Proxy-Authenticate header.

Also other sections of the bis draft need to be updated inline with
this change;

1) Table 5, pg 36, add 401 to "where" column for Proxy-Authenticate
   header.

2) Add note to section 6.31 Proxy Authenticate, discussing use of
   the header in a 401 response in the forking proxy scenario.

What do you think?

Regards
Phil

--
Phil Hoffer                         Ubiquity Software Corporation
Software Developer                  Ubiquity House
Tel: +44 (0) 1633-765600            Langstone Park
Fax: +44 (0) 1633-765601            Newport   NP18 2LH
URL: http://www.ubiquity.net




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From sip-admin@lists.bell-labs.com  Tue May 23 10:56:17 2000
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From: "Alex Hardisty" <alex.hardisty@pqmconsultants.com>
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Cc: "e-tc32-tg17" <e-tc32-tg17@ecma.ch>, "e-tc32-tg14" <e-tc32-tg14@ecma.ch>
Subject: RE: [SIP] SIP ISUP MIME type
Date: Tue, 23 May 2000 15:56:54 +0100
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Lyndon Ong (Nortel) wrote (on the IETF SIP WG mailing list):

> ...are there any other comments on draft-ietf-sip-isup-mime-01.txt?

To my knowledge the list of QSIG versions is incorrect. At present there are
3 different versions of QSIG. I have listed them below, each followed by my
proposed value for the version parameter:

- the first European version (1992)         : euro
- the first International version (1995)    : iso
- the internationally aligned version (1997): global

I assume that it is possible to extend the list in the future if necessary.
It would be useful, for example, to be able to define further versions (for
example, "globalN" where N is a digit indicating successive versions) should
that prove necessary. Note that I am neither stating nor advocating that
there will be future versions ... just catering for the possibility!!

--
Alex Hardisty
PQM Consultants



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From: "Sinnreich, Henry" <Henry.Sinnreich@WCOM.Com>
Subject: RE: [SIP] Service/subscriber information definitions
To: "'Graham Klyne'" <GK@Dial.pipex.com>,
        James Kempf <James.Kempf@eng.sun.com>
Cc: mat@cisco.com, mramalho@cisco.com, giuseppe.ricagni@italtel.it,
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>There has been much talk over recent months of the Internet's 
>end-to-end 
>architecture, and the way that it allows innovation by any 
>connected party 
>rather than just those that control the core. 

Yes, this is a key issue. Maybe we should be aware what is 'fogging the
internet':
 * Firewalls,
 * NATS
 * WAP (discussion at 47th IETF plenary)
 * Web load balancing switching and web caching
 * Generic "softswitches" with proprietary Call Agent - similar Garden Wall
to services like WAP.

Would actually appreciate the feelings of the list members on the effects of
"soft switches" on Internet Transparency (RFC 2775).

Thanks, Henry

>-----Original Message-----
>From: Graham Klyne [mailto:GK@Dial.pipex.com]
>Sent: Tuesday, May 23, 2000 3:44 AM
>To: James Kempf
>Cc: mat@cisco.com; mramalho@cisco.com; giuseppe.ricagni@italtel.it;
>sip@lists.bell-labs.com
>Subject: [SIP] Service/subscriber information definitions
>
>
>At 01:41 PM 5/22/00 -0700, James Kempf wrote:
>
>> >However, it is *very* clear to me that in the "new world"
>> >of web-based or web-like services (and services definition)
>> >that standardization of anything but the most rudimentary
>> >subscriber information or core services will not make it in
>> >"new world" deployments.
>> >
>> >Don't fence me in,
>> >
>>
>>Don't see why this should be any more the case for a telephony
>>DTD than for any other standardized XML document. Naturally,
>>there needs to be some provision in the format for extensibility,
>
>[...]
>
>There has been much talk over recent months of the Internet's 
>end-to-end 
>architecture, and the way that it allows innovation by any 
>connected party 
>rather than just those that control the core.  (I think SIP is 
>just one 
>fruit of this realization.)
>
>I think there is a new front emerging over which the tensions between 
>centralized and decentralized control will occur:  content 
>formats.  How 
>many organizations are stuck with a certain vendor's software 
>because their 
>documents are in the format of that software?  Through XML, 
>XML namespaces, 
>RDF and other initiatives, W3C are working on content formats 
>that offer 
>the potential for an end-to-end content architecture, much as IP has 
>provided an end-to-end communication architecture.  The key is 
>something 
>that Tim Berners-Lee has called "language mixing".  (see 
>http://www.w3.org/DesignIssues/Evolution.html)
>
>These ideas are being developed in the context of WWW, but I 
>belive are 
>applicable to other Interbnet applications.
>
>In summary:  using the right tools, a definition of core 
>information should 
>not be a fence, but a springboard.  (In this, I am broadly 
>agreeing with 
>James.)
>
>#g
>
>
>------------
>Graham Klyne
>(GK@ACM.ORG)
>
>
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip
>


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James Kempf writes:
 > Right. The other thing a standardized user profile format would let me do is 
 > edit services on any  Web browser rather than have to call up the phone company 
 > and wait for them change my service profile. In New World Telephony,
 > this empowers the user so that they effectively control their services.
 > The phone company benefits in that their administrative costs are reduced.
 > And if the XML DTD for the user profile is made extensible, phone
 > service providers can add customized features, just like Web pages
 > have different features.

   If you don't have an XML description of a
   subscriber's record you can't edit it in a web
   browser? Are you suggesting that Sun's and 
   Oracle have been selling web gear for the last
   few years no particular reason at all? :-)

   Seriously, XML descriptions of user data might
   be nice if they are used wisely (cf Jonathan's
   comments), but they don't take the place of 
   graphic design, UI's etc, etc. My point, if it's
   still unclear, is that forcing standardization 
   to be a  _prerequisite_ to deploying service is
   likely to hamstring innovation if multiple 
   service providers need to agree on anything.
   I have no such third party dependencies when
   it's just between me and my service provider.
   It might be nice if and when they got their
   act together, but I don't want to depend on
   that for my cool new features.

	     Mike


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From: "Mo Zonoun" <mzonoun@nortelnetworks.com>
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Subject: RE: [SIP] SIP ISUP MIME type
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Alex, I was under the impression that once ISO standards are developed
national variation are withdrawn. Do we really want to encourage each
country or region have their own specific version? If that is the case why
bother with ISO?

Mo Zonoun 

-----Original Message-----
From: Alex Hardisty [mailto:alex.hardisty@pqmconsultants.com]
Sent: Tuesday, May 23, 2000 7:57 AM
To: sip
Cc: e-tc32-tg17; e-tc32-tg14
Subject: RE: [SIP] SIP ISUP MIME type


Lyndon Ong (Nortel) wrote (on the IETF SIP WG mailing list):

> ...are there any other comments on draft-ietf-sip-isup-mime-01.txt?

To my knowledge the list of QSIG versions is incorrect. At present there are
3 different versions of QSIG. I have listed them below, each followed by my
proposed value for the version parameter:

- the first European version (1992)         : euro
- the first International version (1995)    : iso
- the internationally aligned version (1997): global

I assume that it is possible to extend the list in the future if necessary.
It would be useful, for example, to be able to define further versions (for
example, "globalN" where N is a digit indicating successive versions) should
that prove necessary. Note that I am neither stating nor advocating that
there will be future versions ... just catering for the possibility!!

--
Alex Hardisty
PQM Consultants



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charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2651.65">
<TITLE>RE: [SIP] SIP ISUP MIME type</TITLE>
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<BODY>

<P><FONT SIZE=3D2>Alex, I was under the impression that once ISO =
standards are developed national variation are withdrawn. Do we really =
want to encourage each country or region have their own specific =
version? If that is the case why bother with ISO?</FONT></P>

<P><FONT SIZE=3D2>Mo Zonoun </FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Alex Hardisty [<A =
HREF=3D"mailto:alex.hardisty@pqmconsultants.com">mailto:alex.hardisty@pq=
mconsultants.com</A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Tuesday, May 23, 2000 7:57 AM</FONT>
<BR><FONT SIZE=3D2>To: sip</FONT>
<BR><FONT SIZE=3D2>Cc: e-tc32-tg17; e-tc32-tg14</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [SIP] SIP ISUP MIME type</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Lyndon Ong (Nortel) wrote (on the IETF SIP WG mailing =
list):</FONT>
</P>

<P><FONT SIZE=3D2>&gt; ...are there any other comments on =
draft-ietf-sip-isup-mime-01.txt?</FONT>
</P>

<P><FONT SIZE=3D2>To my knowledge the list of QSIG versions is =
incorrect. At present there are</FONT>
<BR><FONT SIZE=3D2>3 different versions of QSIG. I have listed them =
below, each followed by my</FONT>
<BR><FONT SIZE=3D2>proposed value for the version parameter:</FONT>
</P>

<P><FONT SIZE=3D2>- the first European version =
(1992)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : euro</FONT>
<BR><FONT SIZE=3D2>- the first International version =
(1995)&nbsp;&nbsp;&nbsp; : iso</FONT>
<BR><FONT SIZE=3D2>- the internationally aligned version (1997): =
global</FONT>
</P>

<P><FONT SIZE=3D2>I assume that it is possible to extend the list in =
the future if necessary.</FONT>
<BR><FONT SIZE=3D2>It would be useful, for example, to be able to =
define further versions (for</FONT>
<BR><FONT SIZE=3D2>example, &quot;globalN&quot; where N is a digit =
indicating successive versions) should</FONT>
<BR><FONT SIZE=3D2>that prove necessary. Note that I am neither stating =
nor advocating that</FONT>
<BR><FONT SIZE=3D2>there will be future versions ... just catering for =
the possibility!!</FONT>
</P>

<P><FONT SIZE=3D2>--</FONT>
<BR><FONT SIZE=3D2>Alex Hardisty</FONT>
<BR><FONT SIZE=3D2>PQM Consultants</FONT>
</P>
<BR>
<BR>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>SIP mailing list</FONT>
<BR><FONT SIZE=3D2>SIP@lists.bell-labs.com</FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://lists.bell-labs.com/mailman/listinfo/sip" =
TARGET=3D"_blank">http://lists.bell-labs.com/mailman/listinfo/sip</A></F=
ONT>
</P>

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James Kempf [mailto:James.Kempf@eng.sun.com] writes:
  The other thing a standardized user profile format would let me do is
  edit services on any  Web browser rather than have to call up the phone 
  company and wait for them change my service profile. 

That may be true, but a standardized user profile format certainly isn't
required to be able to edit services with a web browser!  
	
Tim Schroeder
tschroeder@tri.sbc.com
SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759


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From: "Sinnreich, Henry" <Henry.Sinnreich@WCOM.Com>
Subject: RE: [SIP] Minutes of SIP Security task force in Adelaide
To: "'Schroeder, Tim'" <schroeder@tri.sbc.com>, sip@lists.bell-labs.com
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>a standardized user profile format 
>certainly isn't
>required to be able to edit services with a web browser! 

There may be outsourcing arrangements between service providers or with
resellers, where a standard user profile format is beneficial. The customer
may be an enterprise with thousands of individual users. If a change is
required in the OSS, entering it by hand in a web browser is a scary
proposition.

This is the reason why I believe the user profile should be machine
readable, in standard XML format.

Henry

>-----Original Message-----
>From: Schroeder, Tim [mailto:schroeder@tri.sbc.com]
>Sent: Tuesday, May 23, 2000 11:13 AM
>To: sip@lists.bell-labs.com
>Subject: RE: [SIP] Minutes of SIP Security task force in Adelaide
>
>
>James Kempf [mailto:James.Kempf@eng.sun.com] writes:
>  The other thing a standardized user profile format would let me do is
>  edit services on any  Web browser rather than have to call 
>up the phone 
>  company and wait for them change my service profile. 
>
>That may be true, but a standardized user profile format 
>certainly isn't
>required to be able to edit services with a web browser!  
>	
>Tim Schroeder
>tschroeder@tri.sbc.com
>SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759
>
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip
>


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From sip-admin@lists.bell-labs.com  Tue May 23 13:22:26 2000
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I've made changes in various places reflecting this discussion. Thanks
for the comments.

Phil Hoffer wrote:
> 
> Hi,
> 
> IMHO various sections within the new version of the bis draft should
> be updated to reflect the changes made to the paragraph dealing
> with 4xx,5xx response in section 12.4, pg 84.
> 
> The modification deals with how a forking proxy should handle multiple
> authentication responses.
> 
> This modification means that SIP authentication moves further away
> from the process defined in rfc2617, HTTP Authentication: Basic and
> Digest Access Authentication.
> 
> Perhaps section 14.1 in the bis draft should include a list of
> general differences w.r.t. SIP Authentication.
> 
> The topics I'd like to see mentioned are:
> 
> 1) Now possible for user agent to receive challenges from more than
>    1 realm in a single response. This contradicts last paragraph of
>    section 3.6 in rfc2617.
> 
> 2) Possible to receive Proxy-Authenticate header in a 401 response.
> 
>    That said, it's not explicitly stated in rfc2617 that a 401 response
>    can't contain a Proxy-Authenticate header.
> 
> Also other sections of the bis draft need to be updated inline with
> this change;
> 
> 1) Table 5, pg 36, add 401 to "where" column for Proxy-Authenticate
>    header.
> 
> 2) Add note to section 6.31 Proxy Authenticate, discussing use of
>    the header in a 401 response in the forking proxy scenario.
> 
> What do you think?

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


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From sip-admin@lists.bell-labs.com  Tue May 23 14:14:29 2000
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Subject: [SIP] REGISTER/OPTIONS and Call-ID
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In the bis draft, 6.12 Call-ID, it says....

"The REGISTER and OPTIONS methods use the Call-ID to unambiguously match
requests to responses."

In a REGISTER the Call-ID SHOULD remain the same over at least the same
reboot cycle. So isn't the CSeq also required to match requests to
responses? If not, why MUST registrations with the same Call-ID have
increasing CSeq values?

The bis draft also isn't explicity clear to me as to what the Call-ID of
OPTIONS should be. Should the Call-ID of an OPTIONS be analogous to an
INVITE not the special case of a REGISTER?

Cheers,
Neil.
--
Ubiquity Software Corporation           http://www.ubiquity.net



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Neil Deason wrote:
> 
> In the bis draft, 6.12 Call-ID, it says....
> 
> "The REGISTER and OPTIONS methods use the Call-ID to unambiguously match
> requests to responses."
> 
> In a REGISTER the Call-ID SHOULD remain the same over at least the same
> reboot cycle. So isn't the CSeq also required to match requests to
> responses? If not, why MUST registrations with the same Call-ID have
> increasing CSeq values?

Added "(together with CSeq)" for clarity.

> 
> The bis draft also isn't explicity clear to me as to what the Call-ID of
> OPTIONS should be. Should the Call-ID of an OPTIONS be analogous to an
> INVITE not the special case of a REGISTER?

Either:

"For these [OPTIONS, REGISTER] requests, it makes no difference whether
the
\header{Call-ID} value matches an existing call or not."


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


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From sip-admin@lists.bell-labs.com  Tue May 23 17:25:33 2000
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From: "Schroeder, Tim" <schroeder@tri.sbc.com>
To: sip@lists.bell-labs.com
Subject: RE: [SIP] Carrying Q.BICC as an Mime-type in SIP
Date: Tue, 23 May 2000 16:24:30 -0500
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Mark Watson [mailto:mwatson@nortelnetworks.com] writes:

 If we have CS(A) --bicc ---CS(B)---sip --- CS(C)---bicc---CS(D) 

where 'CS' is 'Call Server', for want of a better term, then there are two
cases: 

(i) with the bearer path routed directly between Gateways controlled by
CS(A) and CS(D), and 

(ii) with further 'packet-to-packet' gateways contolled by CS(B) and CS(C)
i.e. there are three 'domains' which are physically interworked at both the
call and bearer layer.

There is some doubt as to whether (i) will be possible at all - this depends
on how the support of IP in BICC is defined over the next few months. If it
is possible, I think that it should involve maping the bearer control
mechanism in BICC to the equivalent in SIP (SDP) so that the call can still
work if it terminates on a pure SIP agent. 

Agreed.

If you do this mapping, then you've removed from the BICC information
everything which makes it BICC and not ISUP, so you're not carrying BICC
inside SIP at all. 

But just because you map the information from BICC into SIP, doesn't mean
it's removed from the BICC message, does it?  In the ISUP--SIP--ISUP case,
the ISUP information that makes sense to map to SIP (to provide the desired
level of interworking) is mapped, but the ISUP message is still passed along
in the body to allow for ISUP transparency, in case the call ends up back on
an ISUP-signaled network.  I'd think the same mechanism would apply to the
BICC--SIP--BICC case: map the BICC information that makes sense to SIP (and
the bearer stuff to SDP), but pass along the original BICC message in the
body.  Then if the call terminates in the SIP-signaled network, you've got
your interworking (to the level that you coherently mapped the BICC
information to SIP/SDP), and if the call is routed through a gateway back to
a BICC-signaled network you can still have "BICC transparency".

In case (ii), the three domains are completely independent from each other
when it comes to setup of the bearer connection. Again, the only information
from BICC left to carry over the SIP domain is the ISUP part. 

I think I'd still want to send the whole BICC message, even if the non-ISUP
part will never be used.  The gateways provide the interworking
functionality between BICC and SIP; there's no need to have them also
interwork between BICC and ISUP (even if that's relatively easy) just so I
can avoid an application/BICC or application/Q.1901 MIME type. 

If we were to carry the BICC APM parameter, which contains the Bearer
related information, over SIP, we would be inventing a new way of carrying
out bearer setup on a SIP call - an alternative to SDP. I do not think we
should do this as it would be non-backwards-compatible. 

In your case (i), interwork the bearer related information to SDP (which
will get used if the call terminates in the SIP-signaled network, ignoring
the BICC-embedded APM parameter) and leave it in the BICC message as well,
passing the whole BICC message along in the body (which will get used if the
call is routed back to the BICC-signaled network).  

In your case (ii), the gateways are going to have to piece together the
three media streams end-to-end, a first BICC APM parameter will describe the
CS(A)--CS(B) segment, SDP will describe the CS(B)--CS(C) segment, and a
second BICC APM parameter will describe the CS(C)--CS(D) segment.  I don't
think there's much BICC/APM-to-SDP interworking that can be done there, but
the BICC message may as well be passed along, rather than first converting
it to the ISUP format.

Tim Schroeder 
tschroeder@tri.sbc.com 
SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759 

 


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From sip-admin@lists.bell-labs.com  Tue May 23 17:43:20 2000
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Dear Sir,
	I need some technecal information about SIP.How it works,what are
the hardware requirments to install sip in my I.I.T. Lab.Sir please give 
me required information at [skamit@iitk.ac.in]
					Amit Kumar Singh
					I.I.T. Kanpur
					INDIA



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From sip-admin@lists.bell-labs.com  Tue May 23 18:35:28 2000
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From: "Schroeder, Tim" <schroeder@tri.sbc.com>
To: sip@lists.bell-labs.com
Subject: RE: [SIP] Comments on Session Timer draft
Date: Tue, 23 May 2000 17:34:40 -0500
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I was reading some of the old messages commenting on the session timer
draft, and there was a long discussion about whether re-INVITE or some other
method was the "correct" method to use to refresh the timer.  A re-INVITE
might set up a session that had fallen down, which was perceived to be good.
There were issues with Cseq space (addressed by basing them on a local
clock), routes (addressed by proxies adding Record-Route to every request
they see, media intolerant of sessions being set up again without being
cleanly torn down first, and tags.  (I didn't see anything addressing the
last two.)

I'm wondering, though, why the feature must require any specific method at
all.  If the user agent wants the behavior of re-INVITE, it could send
re-INVITEs to refresh the timer.  If it doesn't it could send an INFO.  In
fact, if it were sending "real" re-INVITES anyway (to adjust the media),
there'd be no need to send another one just to refresh the timer.  Or if it
were sending INFO messages anyway (perhaps to keep track of some billing
stuff), there'd be no need to send any sort of re-INVITE or INFO just to
refresh the timer.  And if the user agent were *receiving* messages of any
type from the far end, it would also be able to skip the timer refresh
messages.

Maybe the draft could say that *any* message (sent or received) would
refresh the timer?  And if the timer were about to expire, that only then
must a message be sent?  And it could suggest any of a re-INVITE, or an
INFO, or ...  -- the exact message type wouldn't matter, only that one was
sent or received.

Tim Schroeder
tschroeder@tri.sbc.com
SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759



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From sip-admin@lists.bell-labs.com  Tue May 23 21:55:56 2000
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http://www.cs.columbia.edu/~hgs/sip

Amit Kumar Singh tr cc wrote:
> 
> Dear Sir,
>         I need some technecal information about SIP.How it works,what are
> the hardware requirments to install sip in my I.I.T. Lab.Sir please give
> me required information at [skamit@iitk.ac.in]
>                                         Amit Kumar Singh
>                                         I.I.T. Kanpur
>                                         INDIA
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Wed May 24 03:15:27 2000
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Subject: RE: [SIP] SIP ISUP MIME type
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L. Ong wrote:
>Folks,
>
>Apart from Kevin's comment on explicitly specifying the starting boundary
>for an encapsulated ISUP message, are there any other comments on
>draft-ietf-sip-isup-mime-01.txt?  
>

In the Table 1 in 3.1, is it possible to add Japanese ISUP variants on the list?

If one of the objectives of "base" parameter is to save the namespace for
each local standard body, the base name for the Japanese variants should be 
designated when SIP-T is applied to Japanese network.
I feel it is better to be designated in RFC of ISUP MIME type than in any Japanese
local documents.

I do not think it is a good idea to list the all variants here and also I know 
"Table 1 is a partial list". But I feel Japanese variants are major and different
enough to be put in the list.

If you think it is not a bad idea, I think I can discuss this with the members 
of TTC, which has the responsibility for the Japanese telecommunications 
standardization, and tell you what is the adequate base name for Japanese variants.

Regards,
Takuya Sawada



------
Takuya SAWADA (tu-sawada@kdd.co.jp)
 Network Development Div. Switcing System Dept.
 IP telephone Group
 2-3-2 Nishishinjuku Shinjuku-ku, Tokyo 163-8003, Japan
 Tel:+81 3 3347 5663    Fax:+81 3 3347 5826



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From sip-admin@lists.bell-labs.com  Wed May 24 03:21:05 2000
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L. Ong wrote:
>Folks,
>
>Apart from Kevin's comment on explicitly specifying the starting boundary
>for an encapsulated ISUP message, are there any other comments on
>draft-ietf-sip-isup-mime-01.txt?  
>

Regarding 'version' parameter, the last sentence of the first paragraph on
page 3 says;

  "If not specified, the default version is assumed to be ITU-T ISUP 1992".

But 'version' parameter is defined as a required parameter in the draft.
And also NOTE says "It is mandatory for client to specify the 'version' of
the ISUPmessage".
So the sentence above is correct?

Regarding 'base' parameter, it is an optional parameter.
Should we define the default value for 'base' parameter, such as 
"the default base is assumed to be ITU-T ISUP 1992, that is 'base=itu-t92+'" ?

Regards,
Takuya Sawada



------
Takuya SAWADA (tu-sawada@kdd.co.jp)
 Network Development Div. Switcing System Dept.
 IP telephone Group
 2-3-2 Nishishinjuku Shinjuku-ku, Tokyo 163-8003, Japan
 Tel:+81 3 3347 5663    Fax:+81 3 3347 5826



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Subject: RE: [SIP] SIP ISUP MIME type
Date: Wed, 24 May 2000 09:37:30 +0100
Message-ID: <NDBBLFKHOLNEHLCNJANACEBMCCAA.alex.hardisty@pqmconsultants.com>
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Mo Zonoun wrote:

> I was under the impression that once ISO standards are developed
> national variation are withdrawn. Do we really want to encourage
> each country or region have their own specific version? If that is
> the case why bother with ISO?

That is true and I wasn't suggesting anything different.

From the point of view of transporting QSIG information in SIP messages,
however, there are implementations of each of the variants in the field and
if it is necessary to indicate the version being transported then this
should be done correctly. The current Internet Draft (I-D) lists two
variants; neither of which (IMHO) accurately describes the actual situation.

Regards
--
Alex Hardisty
PQM Consultants



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From: Linden deCarmo <ldeCarmo@netspeak.com>
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Subject: RE: [SIP] Comments on Session Timer draft
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	>Maybe the draft could say that *any* message (sent or received)
would
>refresh the timer? 

I think this is too open ended.  Since it is possible to create user-defined
methods, this would imply that a server implementing the session-timer
support would have to parse all packets, regardless if it understood the
method or not.  In addition, how could you return a 501 (Not Implemented) if
you were looking for and using a header in the packet?

I much prefer standardizing on a specific method.  Greatly simplifies the
implementation. 

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



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From sip-admin@lists.bell-labs.com  Wed May 24 10:15:28 2000
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Subject: Re: [SIP] Comments on Session Timer draft
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I agree. Lets keep it simple. There are clear benefits to using
re-INVITE, as we have discussed in the past.

-Jonathan R.

Linden deCarmo wrote:
> 
>         >Maybe the draft could say that *any* message (sent or received)
> would
> >refresh the timer?
> 
> I think this is too open ended.  Since it is possible to create user-defined
> methods, this would imply that a server implementing the session-timer
> support would have to parse all packets, regardless if it understood the
> method or not.  In addition, how could you return a 501 (Not Implemented) if
> you were looking for and using a header in the packet?
> 
> I much prefer standardizing on a specific method.  Greatly simplifies the
> implementation.
> 
> Linden deCarmo
> Netspeak Corporation
> 902 Clint Moore Road
> Suite 104
> Boca Raton, FL 33487
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Wed May 24 10:46:55 2000
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hi, all,

If a UAC sends out an INVITE, which will tranverse 4 proxy servers until it
reaches the callee, the servers provide different services, this is standard
method of SIP. 

There is another approach: UAC sends out an INVITE to the first proxy
server, which may be its local proxy, then the proxy acts as a client and
sends the different service requests to the other 3 proxy servers. Upon the
first server receives the responses, it forwards the INVITE directly to the
callee, so the INVITE only need to tranverse ONE proxy.

Is that possible?

thanks for all reply.

Yu



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From sip-admin@lists.bell-labs.com  Wed May 24 11:43:10 2000
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Subject: [SIP] Options Tags - v.minor change req'd to bis text
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This doesn't effect protocol operation in anyway but Option Tags
(Section 4.4) of bis draft reads ...

"These tags are used in Require (Section 6.35) and Unsupported (Section
6.44) fields."

With the addition of Supported in the bis draft that should now read ...

"These tags are used in Require (Section 6.35), Supported (Section 6.41)
and Unsupported (Section 6.44) fields."

hth,
Neil
--
Ubiquity Software Corporation           http://www.ubiquity.net




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From sip-admin@lists.bell-labs.com  Wed May 24 12:10:09 2000
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This draft states that a probing mechanism should be defined to aid
backwards compatibility when defining new methods. It describes the
mechanism to be used as the UAC adding some header to a standard SIP
request. The UAS places some information in the response if it
understands this header (and thus the extension-method).

Couldn't the OPTIONS method be used to provide such a probing mechanism
within the standard SIP framework? The response to an OPTIONS request
should include an Allow header entity-field that lists the supported set
of headers, including any extension-methods.

Cheers,
Neil.
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From sip-admin@lists.bell-labs.com  Wed May 24 12:28:21 2000
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To: "Dean Willis" <dean.willis@softarmor.com>,
        "Michael Thomas" <mat@cisco.com>
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Subject: RE: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task force in Adelaide)
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If you're using Intserv at the edge the router at the edge of the Diffserv
cloud will likely know this and reject the reservation. No need for the
proxy to get involved.

The car analogy is not a good one because in the case of highways you're
already stuck at the bottleneck when you find out you're screwed, as opposed
to RSVP when you are essentailly clearing all the dead wrecks out of your
way before driving up the highway, ala Mad Max.

Positing that a SIP proxy has a better idea what the QoS state of the
Diffserv core is than the edge router at the edge of that core is like
saying the radio announcer knows more about the tornado than the Twister
crew chasing it.

If there's a oracle which knows the state of the core it can tell the edges
as easily or MORE easily than it can tell an undefined sea of application
proxy boxes.

Dave.

> -----Original Message-----
> From: sip-admin@lists.bell-labs.com
> [mailto:sip-admin@lists.bell-labs.com]On Behalf Of Dean Willis
> Sent: Friday, May 19, 2000 6:35 PM
> To: Michael Thomas
> Cc: IETF SIP
> Subject: RE: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task
> force in Adelaide)
>
>
>
> Ah. The networkstatus-aware proxy is effectively advising the SIP
> user of congestion in the DiffServ network. Kind of like those
> radio announcers that say "Highway 18 is shut down again --
> please use surface streets or just stay home".
>
> The real question is to define a mechanism that allows the health
> of the network to be assessed in some sort of meaningful way so
> that this knowledge may be made available to the UAC.
>
> --
> Dean
>
>
>
> > -----Original Message-----
> > From: Michael Thomas [mailto:mat@cisco.com]
> > Sent: Thursday, May 18, 2000 4:32 PM
> > To: Dean Willis
> > Cc: IETF SIP; Michael Thomas
> > Subject: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task force
> > in Adelaide)
> >
> >
> > Dean Willis writes:
> >  > Consider it in an aggregation model, like a donut. The chewy
> > outside parts of the donut are controlled by IntServ. Traffic
> > leaps across the DiffServ donut-hole only if the heuristic
> > process thinks (IE, is reasonably sure) that there is adequate
> bandwidth.
> >
> >    OK, but I still don't see what that has to do
> >    with proxies. That sounds like something an
> >    edge router would do before it admits diffserv
> >    traffic into the diffserv core. Proxies are even
> >    more clueless because they never see the actual
> >    traffic patterns.
> >
> > 		Mike
> > Hƒæj)bž	b²Ôˆ>X¬¶ÆÞ–YZnÇ(šm§ÿåŠËlmée•¦ìr‰¿™¨¥™©ÿ–+-ŠwèþÈ©



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From: "Holland, Peter Michael (Peter)" <pmholland@lucent.com>
To: sip@lists.bell-labs.com, "'Schroeder, Tim'" <schroeder@tri.sbc.com>
Subject: RE: [SIP] Carrying Q.BICC as an Mime-type in SIP
Date: Wed, 24 May 2000 17:23:32 +0100
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Considering Tim Schroeder's response to Mark Watson's mail, it was
a little difficult to distinguish between the original, and the response,
but
concerning these extracts:

> In case (ii), the three domains are completely independent from each other
> when it comes to setup of the bearer connection. Again, the only
> information
> from BICC left to carry over the SIP domain is the ISUP part. 
> 
> I think I'd still want to send the whole BICC message, even if the
> non-ISUP
> part will never be used.  The gateways provide the interworking
> functionality between BICC and SIP; there's no need to have them also
> interwork between BICC and ISUP (even if that's relatively easy) just so I
> can avoid an application/BICC or application/Q.1901 MIME type. 
> 
....

> In your case (ii), the gateways are going to have to piece together the
> three media streams end-to-end, a first BICC APM parameter will describe
> the
> CS(A)--CS(B) segment, SDP will describe the CS(B)--CS(C) segment, and a
> second BICC APM parameter will describe the CS(C)--CS(D) segment.  I don't
> think there's much BICC/APM-to-SDP interworking that can be done there,
> but
> the BICC message may as well be passed along, rather than first converting
> it to the ISUP format.
> 
The way the BICC will support IP bearers has not yet been finalised, but
this
case (ii) is setting up the bearers leg-by-leg, as in BICC CS1. The
following
should thus be applicable:  The information in a BICC APP parameter is of 2
types:
1. that which relates to this bearer segment.
2. that which relates to the codec negotiation procedure.

The first of these is not relevant on a subsequent leg of the call
and should not be passed on as it is likely to cause confusion on a
subsequent leg.

The concern expressed refers to "converting to the ISUP format".  I'm not
sure
what the concern really is, considering that the BICC message format is the
same
as the ISUP message format, from the message type octet onwards.


    Pete  Holland  
     Lucent Technologies,  Malmesbury, UK
     email: pmholland@lucent.com
     phone:  +44 1666 832423
     fax:  +44 1666 824515


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From sip-admin@lists.bell-labs.com  Wed May 24 13:25:39 2000
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There was a recent discussion on this list (CANCEL related questions)
about the validity of the opening line of Section 12.4 Forking Proxy
...

"The server must respond to the request immediately with a 100 (Trying)
response."

I think the general consensus was that this is too sweeping a statement.
In the case of a CANCEL request this doesn't make sense as the Proxy
will respond immediately, probably with 200 but possibly 481. It also
makes no sense for an ACK.

It seems worth repeating the 200ms rule of Section 7.1 here ...

"The server must respond to the request immediately with a 100 (Trying)
response if it expects to take more than 200ms to obtain a final
response."

Cheers,
Neil.
--
Ubiquity Software Corporation, UK           http://www.ubiquity.net




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From sip-admin@lists.bell-labs.com  Wed May 24 14:24:02 2000
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Ge Yu wrote:
> 
> hi, all,
> 
> If a UAC sends out an INVITE, which will tranverse 4 proxy servers until it
> reaches the callee, the servers provide different services, this is standard
> method of SIP.
> 
> There is another approach: UAC sends out an INVITE to the first proxy
> server, which may be its local proxy, then the proxy acts as a client and
> sends the different service requests to the other 3 proxy servers. Upon the
> first server receives the responses, it forwards the INVITE directly to the
> callee, so the INVITE only need to tranverse ONE proxy.

I think you are talking about forking, but I'm not certain. SIP allows a
proxy to receive one request, and forward more than one request out. IN
this case, the local proxy would fork and send three INVITEs. These
would hit three separate servers, each of which tries to contact the
user. As they receive responses, these responses are forwarded upstream
to the forking proxy. The first call acceptance that the forking proxy
receives is forwarded upstream towards the caller.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Wed May 24 15:53:50 2000
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Dave Oran wrote:
> 
> If you're using Intserv at the edge the router at the edge of the Diffserv
> cloud will likely know this and reject the reservation. No need for the
> proxy to get involved.

In general, I'm not. The WorldCom net, for example, is TOS-priority from
end to end.

> 
> The car analogy is not a good one because in the case of highways you're
> already stuck at the bottleneck when you find out you're screwed, as opposed
> to RSVP when you are essentailly clearing all the dead wrecks out of your
> way before driving up the highway, ala Mad Max.
> 
> Positing that a SIP proxy has a better idea what the QoS state of the
> Diffserv core is than the edge router at the edge of that core is like
> saying the radio announcer knows more about the tornado than the Twister
> crew chasing it.

A SIP proxy is in a better position to have a persistent relationship
with a network management system which can feed it information about the
state of the network than a dynamically romin SIP UA is. So, even though
the announceer may not know as much about the tornado as the chaser
does, he is in a MUCH better position to share that information with the
listening audience.
 
> If there's a oracle which knows the state of the core it can tell the edges
> as easily or MORE easily than it can tell an undefined sea of application
> proxy boxes.

Sure, the edges probably know truckloads of cool stuff.

But I have no mechanism for the edges to relay this knowledge to the
UAs.

--
Dean


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From sip-admin@lists.bell-labs.com  Wed May 24 16:08:51 2000
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Ah. I understand. no RSVP == no explicit admission control.
That's the bug. Hijacking SIP proxies to do the job of a bandwidth
allocation and admission control signaling protocol is ugly at best. You'd
need one of these beasts for every application in the universe - rtsp,
quake, you name it.

If you build SIP proxies the way some people build web server farms you'll
have an entirely different scaling problem, which is 100's of SIP proxies
polling a poor overworked network managmeent system trying to figure out
what it knows about the state of the core (or better stated, what it knew
about the core 5 minutes ago when it last gathered enough measurements to
make a guess).

Register me as a skeptic.

Dave.

> -----Original Message-----
> From: Dean Willis [mailto:dwillis@dynamicsoft.com]
> Sent: Wednesday, May 24, 2000 3:53 PM
> To: Dave Oran
> Cc: Dean Willis; Michael Thomas; IETF SIP
> Subject: Re: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task
> force in Adelaide)
>
>
>
>
> Dave Oran wrote:
> >
> > If you're using Intserv at the edge the router at the edge of
> the Diffserv
> > cloud will likely know this and reject the reservation. No need for the
> > proxy to get involved.
>
> In general, I'm not. The WorldCom net, for example, is TOS-priority from
> end to end.
>
> >
> > The car analogy is not a good one because in the case of highways you're
> > already stuck at the bottleneck when you find out you're
> screwed, as opposed
> > to RSVP when you are essentailly clearing all the dead wrecks
> out of your
> > way before driving up the highway, ala Mad Max.
> >
> > Positing that a SIP proxy has a better idea what the QoS state of the
> > Diffserv core is than the edge router at the edge of that core is like
> > saying the radio announcer knows more about the tornado than the Twister
> > crew chasing it.
>
> A SIP proxy is in a better position to have a persistent relationship
> with a network management system which can feed it information about the
> state of the network than a dynamically romin SIP UA is. So, even though
> the announceer may not know as much about the tornado as the chaser
> does, he is in a MUCH better position to share that information with the
> listening audience.
>
> > If there's a oracle which knows the state of the core it can
> tell the edges
> > as easily or MORE easily than it can tell an undefined sea of
> application
> > proxy boxes.
>
> Sure, the edges probably know truckloads of cool stuff.
>
> But I have no mechanism for the edges to relay this knowledge to the
> UAs.
>
> --
> Dean
>
>



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From sip-admin@lists.bell-labs.com  Wed May 24 16:57:09 2000
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From: "Roy, Radhika R, ALARC" <rrroy@att.com>
To: Dave Oran <oran@cisco.com>, Dean Willis <dwillis@dynamicsoft.com>
Cc: Michael Thomas <mat@cisco.com>, IETF SIP <sip@lists.bell-labs.com>
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Hi, Dave and Dean:

Let me try to understand the issues correctly.

A SIP proxy server is an application level entity that deals with the
application layer signaling scheme.

RSVP, DiffServ, etc are the network layer QOS signaling schemes.

So, I believe that it would be better to have a clear separation between the
two although there can be an abstraction for interworking between the two
layers. The question is: How do we this without creating scalability
problems?

In this context, let us analyze the following:

Should the SIP proxy server do the bandwidth allocation and admission
control?

It depends what the abstraction of bandwidth and admission control is.

In true sense, the admission control to the network based on bandwidth and
other performance criteria will fall into the category of the lower network
layer entities because the network layer resource allocation will be done by
them. 

Now the challenge is: How can this lower network layer information be
abstracted (or coupled) to the application layer SIP proxies that are
responsible to set up the call on end-to-end basis between the SIP agents?

Did I miss something?

Best regards,
Radhika R. Roy

> -----Original Message-----
> From:	Dave Oran [SMTP:oran@cisco.com]
> Sent:	Wednesday, May 24, 2000 4:09 PM
> To:	Dean Willis
> Cc:	Michael Thomas; IETF SIP
> Subject:	RE: SIP DiffServe (was RE: [SIP] Minutes of SIP Security
> task force in Adelaide)
> 
> Ah. I understand. no RSVP == no explicit admission control.
> That's the bug. Hijacking SIP proxies to do the job of a bandwidth
> allocation and admission control signaling protocol is ugly at best. You'd
> need one of these beasts for every application in the universe - rtsp,
> quake, you name it.
> 
> If you build SIP proxies the way some people build web server farms you'll
> have an entirely different scaling problem, which is 100's of SIP proxies
> polling a poor overworked network managmeent system trying to figure out
> what it knows about the state of the core (or better stated, what it knew
> about the core 5 minutes ago when it last gathered enough measurements to
> make a guess).
> 
> Register me as a skeptic.
> 
> Dave.
> 
> > -----Original Message-----
> > From: Dean Willis [mailto:dwillis@dynamicsoft.com]
> > Sent: Wednesday, May 24, 2000 3:53 PM
> > To: Dave Oran
> > Cc: Dean Willis; Michael Thomas; IETF SIP
> > Subject: Re: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task
> > force in Adelaide)
> >
> >
> >
> >
> > Dave Oran wrote:
> > >
> > > If you're using Intserv at the edge the router at the edge of
> > the Diffserv
> > > cloud will likely know this and reject the reservation. No need for
> the
> > > proxy to get involved.
> >
> > In general, I'm not. The WorldCom net, for example, is TOS-priority from
> > end to end.
> >
> > >
> > > The car analogy is not a good one because in the case of highways
> you're
> > > already stuck at the bottleneck when you find out you're
> > screwed, as opposed
> > > to RSVP when you are essentailly clearing all the dead wrecks
> > out of your
> > > way before driving up the highway, ala Mad Max.
> > >
> > > Positing that a SIP proxy has a better idea what the QoS state of the
> > > Diffserv core is than the edge router at the edge of that core is like
> > > saying the radio announcer knows more about the tornado than the
> Twister
> > > crew chasing it.
> >
> > A SIP proxy is in a better position to have a persistent relationship
> > with a network management system which can feed it information about the
> > state of the network than a dynamically romin SIP UA is. So, even though
> > the announceer may not know as much about the tornado as the chaser
> > does, he is in a MUCH better position to share that information with the
> > listening audience.
> >
> > > If there's a oracle which knows the state of the core it can
> > tell the edges
> > > as easily or MORE easily than it can tell an undefined sea of
> > application
> > > proxy boxes.
> >
> > Sure, the edges probably know truckloads of cool stuff.
> >
> > But I have no mechanism for the edges to relay this knowledge to the
> > UAs.
> >
> > --
> > Dean
> >
> >
> 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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From sip-admin@lists.bell-labs.com  Wed May 24 17:40:49 2000
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Actually, I think I completely failed to make my point.

I was not suggesting that the proxy do either bandwidth allocation or
admission control. It might make sense in some cases, but that's not
what I'm talking about.

I'm trying to say that a proxy could be a way to get ADVICE about the
state of a core network to devices on the edge -- that it could
RECOMMEND that a client not try to traverse the net when there is a low
probability of success.

In general, I don't expect "controls" to survive hostile users.
Especially admission-control proxies that one can just go around. Bad
idea, don't go there.

Or maybe we need a general "Hey, how's the net doing?" protocol . . .

--
Dean

"Roy, Radhika R, ALARC" wrote:
> 
> Hi, Dave and Dean:
> 
> Let me try to understand the issues correctly.
> 
> A SIP proxy server is an application level entity that deals with the
> application layer signaling scheme.
> 
> RSVP, DiffServ, etc are the network layer QOS signaling schemes.
> 
> So, I believe that it would be better to have a clear separation between the
> two although there can be an abstraction for interworking between the two
> layers. The question is: How do we this without creating scalability
> problems?
> 
> In this context, let us analyze the following:
> 
> Should the SIP proxy server do the bandwidth allocation and admission
> control?
> 
> It depends what the abstraction of bandwidth and admission control is.
> 
> In true sense, the admission control to the network based on bandwidth and
> other performance criteria will fall into the category of the lower network
> layer entities because the network layer resource allocation will be done by
> them.
> 
> Now the challenge is: How can this lower network layer information be
> abstracted (or coupled) to the application layer SIP proxies that are
> responsible to set up the call on end-to-end basis between the SIP agents?
> 
> Did I miss something?
> 
> Best regards,
> Radhika R. Roy
> 
> > -----Original Message-----
> > From: Dave Oran [SMTP:oran@cisco.com]
> > Sent: Wednesday, May 24, 2000 4:09 PM
> > To:   Dean Willis
> > Cc:   Michael Thomas; IETF SIP
> > Subject:      RE: SIP DiffServe (was RE: [SIP] Minutes of SIP Security
> > task force in Adelaide)
> >
> > Ah. I understand. no RSVP == no explicit admission control.
> > That's the bug. Hijacking SIP proxies to do the job of a bandwidth
> > allocation and admission control signaling protocol is ugly at best. You'd
> > need one of these beasts for every application in the universe - rtsp,
> > quake, you name it.
> >
> > If you build SIP proxies the way some people build web server farms you'll
> > have an entirely different scaling problem, which is 100's of SIP proxies
> > polling a poor overworked network managmeent system trying to figure out
> > what it knows about the state of the core (or better stated, what it knew
> > about the core 5 minutes ago when it last gathered enough measurements to
> > make a guess).
> >
> > Register me as a skeptic.
> >
> > Dave.
> >
> > > -----Original Message-----
> > > From: Dean Willis [mailto:dwillis@dynamicsoft.com]
> > > Sent: Wednesday, May 24, 2000 3:53 PM
> > > To: Dave Oran
> > > Cc: Dean Willis; Michael Thomas; IETF SIP
> > > Subject: Re: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task
> > > force in Adelaide)
> > >
> > >
> > >
> > >
> > > Dave Oran wrote:
> > > >
> > > > If you're using Intserv at the edge the router at the edge of
> > > the Diffserv
> > > > cloud will likely know this and reject the reservation. No need for
> > the
> > > > proxy to get involved.
> > >
> > > In general, I'm not. The WorldCom net, for example, is TOS-priority from
> > > end to end.
> > >
> > > >
> > > > The car analogy is not a good one because in the case of highways
> > you're
> > > > already stuck at the bottleneck when you find out you're
> > > screwed, as opposed
> > > > to RSVP when you are essentailly clearing all the dead wrecks
> > > out of your
> > > > way before driving up the highway, ala Mad Max.
> > > >
> > > > Positing that a SIP proxy has a better idea what the QoS state of the
> > > > Diffserv core is than the edge router at the edge of that core is like
> > > > saying the radio announcer knows more about the tornado than the
> > Twister
> > > > crew chasing it.
> > >
> > > A SIP proxy is in a better position to have a persistent relationship
> > > with a network management system which can feed it information about the
> > > state of the network than a dynamically romin SIP UA is. So, even though
> > > the announceer may not know as much about the tornado as the chaser
> > > does, he is in a MUCH better position to share that information with the
> > > listening audience.
> > >
> > > > If there's a oracle which knows the state of the core it can
> > > tell the edges
> > > > as easily or MORE easily than it can tell an undefined sea of
> > > application
> > > > proxy boxes.
> > >
> > > Sure, the edges probably know truckloads of cool stuff.
> > >
> > > But I have no mechanism for the edges to relay this knowledge to the
> > > UAs.
> > >
> > > --
> > > Dean
> > >
> > >
> >
> >
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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From sip-admin@lists.bell-labs.com  Wed May 24 18:08:04 2000
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From: "Schroeder, Tim" <schroeder@tri.sbc.com>
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        sip@lists.bell-labs.com, "Schroeder, Tim" <schroeder@tri.sbc.com>
Subject: RE: [SIP] Carrying Q.BICC as an Mime-type in SIP
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> Considering Tim Schroeder's response to Mark Watson's mail, it was
> a little difficult to distinguish between the original, and 
> the response,

I hate Outlook.  The copy in my "sent items" folder looks beautiful; what
went out and came back on the mailing list looked wretched.  Sorry.

> The way the BICC will support IP bearers has not yet been finalised, but
this
> case (ii) is setting up the bearers leg-by-leg, as in BICC CS1. The
following
> should thus be applicable:  The information in a BICC APP parameter is of
2
> types:
>      1. that which relates to this bearer segment.
>      2. that which relates to the codec negotiation procedure.
> The first of these is not relevant on a subsequent leg of the call
> and should not be passed on as it is likely to cause confusion on a
> subsequent leg.

I agree that the information of type 1 is not relevant.  I'm just a little
uncomfortable having the gateway on the BICC-to-SIP side edit out part of
the BICC message to keep the SIP network happy.  It may edit out part of the
BICC message for its own reasons (hiding the details of the source network?)
but it shouldn't have to do so just because the SIP network can't
encapsulate BICC, but only straight ISUP.  

I don't know of any reason why it *couldn't* edit it out, it just seems
easier to pass along the whole BICC message transparently.  The destination
BICC network can choose to use the parts it thinks are useful, and to ignore
the parts it thinks aren't.  

In the ISUP case, we send the ISUP message transparently, without editing
out the parts known (or suspected) to be useless in the destination network.
I like that model, and it seems easy enough to follow for BICC as well.  And
even if BICC looks a lot like ISUP, and after the useless information has
been removed may look exactly like ISUP, it still seems cleaner to transmit
the BICC as encapsulated BICC, and label it as such with "application/bicc"
or some such, rather than calling it ISUP. 

> The concern expressed refers to "converting to the ISUP format".  I'm not
sure
> what the concern really is, considering that the BICC message format is
the same
> as the ISUP message format, from the message type octet onwards.

Annex E of the Q.1901 document is all about BICC-to-ISUP interworking.  The
differences don't seem substantial, but ... 

If the BICC information is stripped of its BICC-ness and encapsulated as
ISUP, will the destination end even be able to tell that it used to be BICC?
Could that ever be useful?

Tim Schroeder
tschroeder@tri.sbc.com
SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759



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From sip-admin@lists.bell-labs.com  Wed May 24 18:15:01 2000
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From: Jiri Kuthan <kuthan@fokus.gmd.de>
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Dave,

In the DiffServ case, I see these policies to control the edge:

0) no control; the edge trusts end-devices absolutely; end-devices
   will get what they want
	-> not very secure
1) self-control; the edge manages itself; it uses some heuristics to say
   which packet deserves which class
	-> heuristics may fail
2) the edge is polled/controlled explicitly by
   a) end-devices 
	->they have some built-in control support, e.g. RSVP mapped to 
          DiffServe at the edge
   b) proxies on behalf of the end-devices
	->works w/o changes to end-devices
	->application-awareness may be added on top of the
	  admission control, e.g. a SIP/RTSP/quake proxy will be
          able to decide that all video streams from
	  head of department will be assigned high
	  priority

The question: Is 2b an invalid policy? Is it a 
useful policy? And what is actually the difference in terms 
of scalability between 
i) end-devices trying to control/poll the edge [2a]
ii) farm of SIP proxies doing the same on behalf of them 
    (possibly using some kind of aggregation) [2b]?
The scalability bottleneck is the edge in both cases,
isn't it?

Thanks,

Jiri

Dave Oran wrote:
> 
> Ah. I understand. no RSVP == no explicit admission control.
> That's the bug. Hijacking SIP proxies to do the job of a bandwidth
> allocation and admission control signaling protocol is ugly at best. You'd
> need one of these beasts for every application in the universe - rtsp,
> quake, you name it.
> 
> If you build SIP proxies the way some people build web server farms you'll
> have an entirely different scaling problem, which is 100's of SIP proxies
> polling a poor overworked network managmeent system trying to figure out
> what it knows about the state of the core (or better stated, what it knew
> about the core 5 minutes ago when it last gathered enough measurements to
> make a guess).
> 
> Register me as a skeptic.
> 
> Dave.
> 
> > -----Original Message-----
> > From: Dean Willis [mailto:dwillis@dynamicsoft.com]
> > Sent: Wednesday, May 24, 2000 3:53 PM
> > To: Dave Oran
> > Cc: Dean Willis; Michael Thomas; IETF SIP
> > Subject: Re: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task
> > force in Adelaide)
> >
> >
> >
> >
> > Dave Oran wrote:
> > >
> > > If you're using Intserv at the edge the router at the edge of
> > the Diffserv
> > > cloud will likely know this and reject the reservation. No need for the
> > > proxy to get involved.
> >
> > In general, I'm not. The WorldCom net, for example, is TOS-priority from
> > end to end.
> >
> > >
> > > The car analogy is not a good one because in the case of highways you're
> > > already stuck at the bottleneck when you find out you're
> > screwed, as opposed
> > > to RSVP when you are essentailly clearing all the dead wrecks
> > out of your
> > > way before driving up the highway, ala Mad Max.
> > >
> > > Positing that a SIP proxy has a better idea what the QoS state of the
> > > Diffserv core is than the edge router at the edge of that core is like
> > > saying the radio announcer knows more about the tornado than the Twister
> > > crew chasing it.
> >
> > A SIP proxy is in a better position to have a persistent relationship
> > with a network management system which can feed it information about the
> > state of the network than a dynamically romin SIP UA is. So, even though
> > the announceer may not know as much about the tornado as the chaser
> > does, he is in a MUCH better position to share that information with the
> > listening audience.
> >
> > > If there's a oracle which knows the state of the core it can
> > tell the edges
> > > as easily or MORE easily than it can tell an undefined sea of
> > application
> > > proxy boxes.
> >
> > Sure, the edges probably know truckloads of cool stuff.
> >
> > But I have no mechanism for the edges to relay this knowledge to the
> > UAs.
> >
> > --
> > Dean
> >
> >
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jiri Kuthan               Phone: +49-30-3463 7271
GMD FOKUS                 Fax  : +49-30-3463 8271
Kaiserin-Augusta-Allee 31 E-Mail : kuthan@fokus.gmd.de
D-10589 Berlin, Germany   WWW    : http://www.fokus.gmd.de/usr/kuthan
--
Curious about Internet Telephony? 
Check http://www.fokus.gmd.de/glone/projects/ipt today!


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From sip-admin@lists.bell-labs.com  Wed May 24 18:16:57 2000
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From: James Yu <james.yu@neustar.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, vkg@lucent.com
Cc: sip@lists.bell-labs.com
Subject: RE: [SIP] SIP URL clarification for user parameter
Date: Wed, 24 May 2000 17:15:19 -0500
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There are four pieces of information related to a freephone number.  The
first piece is the dialed 800 number (a service number).  The second piece
is the translated POTS (a telephone number).  The third piece is the network
routing number (NRN, a number used for routing in the GSTN) if the telephone
number is not used for routing (e.g., due to number portability).  The
fourth piece is the NP database dip indicator.  The example assumes that the
dialed 800 number results in a ported telephone number; therefore, NP
database dip or process is performed to get the NRN for call routing.
Assume that a rn parameter is used to carry the NRN and a di parameter is
used to carry the NP database dip indicator. 

If the TO field carries the dialed 800 number, then tel will carry the
translated telephone number, rn will carry the NRN and di will carry the dip
indicator.  But if tel carries the dialed 800 number, then rn will carry the
translated telephone number.  This is fine if the telephone number is not
ported or before NP database dip is performed.  But if the telephone number
is a ported number, a NRN will be retrieved and used to route the call.
That NRN must be carried in the rn, and di will be included.  The problem
now is where to carry the translated telephone number when that happens (the
assumption is that rn will be used for routing when both tel and rn are
present).  Can tel be used to carry both the dialed 800 number and the
translated telephone number (appear twice)?

Jonathan's comparison between subscriber phone number/NRN and domain name/IP
address is very close.  But there is one difference.  In GSTN, the
subscriber phone number (a name in the NP environment) may still be needed
at the terminating GSTN switch in order to terminate the call.  The NRN
usually just allow the GSTN to route the call to the terminating switch (one
NRN for all the subscriber numbers served by that switch, which is the case
in the North America).  In some countries, the NRN can point to a specific
line card at the switch.  But in some other countries, the NRN may just
point to the terminating network instead of the terminating switch.  For the
case where the NRN only points to the terminating network, that network will
perform another NP database dip to retrieve a NRN that either points to the
terminating switch or to a line card at the terminating switch.  So the NRN
may be changed while the call is being routed to the terminating switch.  No
matter what, the NRN is used for call routing. 

Although NP is limited to be within a country (in the national domain), the
numbers to be carried in tel and rn should be international numbers as
indicated in many SIP examples.  So if someone dialed a U.S. 800 number, it
would be carried as 1-800-xxx-xxxx in the SIP (country code=1).  If one
dials the international freephone number, the number carried in the SIP
would be 800-xxx...xxxx (country code=800).  Unless the country code can be
indicated, international numbers will avoid confusion.

I'm open to any suggestions as to how to carry the information in the SIP.
They can be generated due to ISUP-SIP interworking or due to the situation
that a SIP server needs to retrieve the NRN for a ported number (e.g.,
PC-to-GSTN call).  Since number portability (NP) is part of the basic call
processing in countries that support NP, SIP may need to specify the call
handling when all or some of the four pieces of information are present in
the SIP message.  But it is apparent that it is the NRN (rn) that is to be
used to select the terminating GW or the next hop SIP server (e.g., TRIP)
when it is present.
 
James

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Friday, May 19, 2000 3:46 PM
> To: vkg@lucent.com
> Cc: sip@lists.bell-labs.com
> Subject: Re: [SIP] SIP URL clarification for user paramater
> 
> 
> 
> 
> "Vijay K. Gurbani" wrote:
> > 
> > Jonathan/Henning:
> > 
> > In RFC2543bis, section 2, 2nd last paragraph on page 17, you say:
> > 
> >    "The user parameter value "np-queried" indicates that 
> the user part
> >     contains a telephone number AND that the number 
> reflects the result
> >     of a query to the local number portability database".
> > 
> > However, in Figure 3 on page 18, the BNF for user-param is:
> > 
> >    user-param = "user="("phone" | "ip")
> > 
> > Should it be modified to:
> > 
> >    user-param = "user="(("phone" | "np-queried") | "ip")
> > 
> > A look at the archives indicate that Jonathan and Marc 
> Petit-Huguenin
> > had a thread on this around May 10, 2000 with Jonathan stating that
> > there was no consensus on if this was the right way to do this.  Any
> > updates?
> 
> 
> None I've seen. I have no problem with the idea of saying 
> that some kind
> of query has been done, but this particular solution is very 
> specific to
> LNP, and I think there is a bigger picture. 
> 
> I would argue that the need for this thing at all arises because the
> telephone network uses equivalently formatted numbers to refer to two
> things - names and actual routing numbers (i.e., addresses). 
> Its sort of
> like what would have happened if IP addresses and domain 
> names were the
> same in syntax and formatting, such that you couldn't tell 
> one from the
> other without checking with some database. The purpose of LNP is much
> like DNS, to translate an abstract name to a real routable 
> address. Now,
> in the world of URLs, the namespace (or abstractly, the address space)
> of the content of the URL is defined by the scheme. If we 
> view "numbers
> as names" and "numbers as addresses" as difference "namespaces", the
> theoretically correct way to indicate that LNP has been done is by
> defining a new URL scheme for the two:
> 
> tel:18005551212 and
> rn:17325551212
> 
> where LNP (or 800 number lookup in this case) is used to translate.
> 
> I am not proposing to do this, since practically speaking, its a pain.
> But I'd like something more generic to indicate the namespace that the
> number fits into. I don't exactly know what or how to do this. Perhaps
> this is really just phone-context, with the context being
> "routing-number" once translated? 
> 
> -Jonathan R.
> > 
> > Thanks,
> > 
> > - vijay
> > --
> > Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com 
> vkg@acm.org
> > IN Architecture and Internet Software Group
> > Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, 
> Rm 1A-418
> > Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 
> 630 713 0184
> > 
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> -- 
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 


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From: "Sinnreich, Henry" <Henry.Sinnreich@WCOM.Com>
Subject: RE: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task for	ce
 in Adelaide)
To: "'Roy, Radhika R, ALARC'" <rrroy@att.com>, Dave Oran <oran@cisco.com>,
        Dean Willis <dwillis@dynamicsoft.com>
Cc: Michael Thomas <mat@cisco.com>, IETF SIP <sip@lists.bell-labs.com>
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Radhika,
We have taken a stab at this in
http://ietf.org/internet-drafts/draft-sinnreich-interdomain-sip-qos-osp-01.t
xt

Let us know what you think of the model in this draft. Your comments will be
appreciated.
This is a good opportunity to discuss, though under a different title. Such
as sip-qos? :-)

Henry

>-----Original Message-----
>From: Roy, Radhika R, ALARC [mailto:rrroy@att.com]
>Sent: Wednesday, May 24, 2000 3:57 PM
>To: Dave Oran; Dean Willis
>Cc: Michael Thomas; IETF SIP
>Subject: RE: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task
>for ce in Adelaide)
>
>
>Hi, Dave and Dean:
>
>Let me try to understand the issues correctly.
>
>A SIP proxy server is an application level entity that deals with the
>application layer signaling scheme.
>
>RSVP, DiffServ, etc are the network layer QOS signaling schemes.
>
>So, I believe that it would be better to have a clear 
>separation between the
>two although there can be an abstraction for interworking 
>between the two
>layers. The question is: How do we this without creating scalability
>problems?
>
>In this context, let us analyze the following:
>
>Should the SIP proxy server do the bandwidth allocation and admission
>control?
>
>It depends what the abstraction of bandwidth and admission control is.
>
>In true sense, the admission control to the network based on 
>bandwidth and
>other performance criteria will fall into the category of the 
>lower network
>layer entities because the network layer resource allocation 
>will be done by
>them. 
>
>Now the challenge is: How can this lower network layer information be
>abstracted (or coupled) to the application layer SIP proxies that are
>responsible to set up the call on end-to-end basis between the 
>SIP agents?
>
>Did I miss something?
>
>Best regards,
>Radhika R. Roy
>
>> -----Original Message-----
>> From:	Dave Oran [SMTP:oran@cisco.com]
>> Sent:	Wednesday, May 24, 2000 4:09 PM
>> To:	Dean Willis
>> Cc:	Michael Thomas; IETF SIP
>> Subject:	RE: SIP DiffServe (was RE: [SIP] Minutes of SIP Security
>> task force in Adelaide)
>> 
>> Ah. I understand. no RSVP == no explicit admission control.
>> That's the bug. Hijacking SIP proxies to do the job of a bandwidth
>> allocation and admission control signaling protocol is ugly 
>at best. You'd
>> need one of these beasts for every application in the 
>universe - rtsp,
>> quake, you name it.
>> 
>> If you build SIP proxies the way some people build web 
>server farms you'll
>> have an entirely different scaling problem, which is 100's 
>of SIP proxies
>> polling a poor overworked network managmeent system trying 
>to figure out
>> what it knows about the state of the core (or better stated, 
>what it knew
>> about the core 5 minutes ago when it last gathered enough 
>measurements to
>> make a guess).
>> 
>> Register me as a skeptic.
>> 
>> Dave.
>> 
>> > -----Original Message-----
>> > From: Dean Willis [mailto:dwillis@dynamicsoft.com]
>> > Sent: Wednesday, May 24, 2000 3:53 PM
>> > To: Dave Oran
>> > Cc: Dean Willis; Michael Thomas; IETF SIP
>> > Subject: Re: SIP DiffServe (was RE: [SIP] Minutes of SIP 
>Security task
>> > force in Adelaide)
>> >
>> >
>> >
>> >
>> > Dave Oran wrote:
>> > >
>> > > If you're using Intserv at the edge the router at the edge of
>> > the Diffserv
>> > > cloud will likely know this and reject the reservation. 
>No need for
>> the
>> > > proxy to get involved.
>> >
>> > In general, I'm not. The WorldCom net, for example, is 
>TOS-priority from
>> > end to end.
>> >
>> > >
>> > > The car analogy is not a good one because in the case of highways
>> you're
>> > > already stuck at the bottleneck when you find out you're
>> > screwed, as opposed
>> > > to RSVP when you are essentailly clearing all the dead wrecks
>> > out of your
>> > > way before driving up the highway, ala Mad Max.
>> > >
>> > > Positing that a SIP proxy has a better idea what the QoS 
>state of the
>> > > Diffserv core is than the edge router at the edge of 
>that core is like
>> > > saying the radio announcer knows more about the tornado than the
>> Twister
>> > > crew chasing it.
>> >
>> > A SIP proxy is in a better position to have a persistent 
>relationship
>> > with a network management system which can feed it 
>information about the
>> > state of the network than a dynamically romin SIP UA is. 
>So, even though
>> > the announceer may not know as much about the tornado as the chaser
>> > does, he is in a MUCH better position to share that 
>information with the
>> > listening audience.
>> >
>> > > If there's a oracle which knows the state of the core it can
>> > tell the edges
>> > > as easily or MORE easily than it can tell an undefined sea of
>> > application
>> > > proxy boxes.
>> >
>> > Sure, the edges probably know truckloads of cool stuff.
>> >
>> > But I have no mechanism for the edges to relay this 
>knowledge to the
>> > UAs.
>> >
>> > --
>> > Dean
>> >
>> >
>> 
>> 
>> 
>> _______________________________________________
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>> SIP@lists.bell-labs.com
>> http://lists.bell-labs.com/mailman/listinfo/sip
>
>
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From sip-admin@lists.bell-labs.com  Wed May 24 19:42:24 2000
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> Or maybe we need a general "Hey, how's the net doing?" protocol . . .
>
Ping? 



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From sip-admin@lists.bell-labs.com  Wed May 24 19:48:11 2000
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From: "Sinnreich, Henry" <Henry.Sinnreich@WCOM.Com>
Subject: RE: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task for	ce
 in Adelaide)
To: "'Dave Oran'" <oran@cisco.com>, Dean Willis <dean.willis@softarmor.com>,
        Michael Thomas <mat@cisco.com>
Cc: IETF SIP <sip@lists.bell-labs.com>
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Agree Dave,

Our model has the edge router doing the RSVP aggregation to DiffServ and
also knowing the sum of all QoS streams. It is up to the policy server and
bandwidth manager to authorize any new application requesting QoS, SIP in
this case. The authorization depends on:
- the clearinghouse AAA,
- the SLS with the backbone,
- available BW in the local domain.

Henry

>-----Original Message-----
>From: Dave Oran [mailto:oran@cisco.com]
>Sent: Wednesday, May 24, 2000 11:28 AM
>To: Dean Willis; Michael Thomas
>Cc: IETF SIP
>Subject: RE: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task
>force in Adelaide)
>
>
>If you're using Intserv at the edge the router at the edge of 
>the Diffserv
>cloud will likely know this and reject the reservation. No need for the
>proxy to get involved.
>
>The car analogy is not a good one because in the case of 
>highways you're
>already stuck at the bottleneck when you find out you're 
>screwed, as opposed
>to RSVP when you are essentailly clearing all the dead wrecks 
>out of your
>way before driving up the highway, ala Mad Max.
>
>Positing that a SIP proxy has a better idea what the QoS state of the
>Diffserv core is than the edge router at the edge of that core is like
>saying the radio announcer knows more about the tornado than 
>the Twister
>crew chasing it.
>
>If there's a oracle which knows the state of the core it can 
>tell the edges
>as easily or MORE easily than it can tell an undefined sea of 
>application
>proxy boxes.
>
>Dave.
>
>> -----Original Message-----
>> From: sip-admin@lists.bell-labs.com
>> [mailto:sip-admin@lists.bell-labs.com]On Behalf Of Dean Willis
>> Sent: Friday, May 19, 2000 6:35 PM
>> To: Michael Thomas
>> Cc: IETF SIP
>> Subject: RE: SIP DiffServe (was RE: [SIP] Minutes of SIP 
>Security task
>> force in Adelaide)
>>
>>
>>
>> Ah. The networkstatus-aware proxy is effectively advising the SIP
>> user of congestion in the DiffServ network. Kind of like those
>> radio announcers that say "Highway 18 is shut down again --
>> please use surface streets or just stay home".
>>
>> The real question is to define a mechanism that allows the health
>> of the network to be assessed in some sort of meaningful way so
>> that this knowledge may be made available to the UAC.
>>
>> --
>> Dean
>>
>>
>>
>> > -----Original Message-----
>> > From: Michael Thomas [mailto:mat@cisco.com]
>> > Sent: Thursday, May 18, 2000 4:32 PM
>> > To: Dean Willis
>> > Cc: IETF SIP; Michael Thomas
>> > Subject: SIP DiffServe (was RE: [SIP] Minutes of SIP 
>Security task force
>> > in Adelaide)
>> >
>> >
>> > Dean Willis writes:
>> >  > Consider it in an aggregation model, like a donut. The chewy
>> > outside parts of the donut are controlled by IntServ. Traffic
>> > leaps across the DiffServ donut-hole only if the heuristic
>> > process thinks (IE, is reasonably sure) that there is adequate
>> bandwidth.
>> >
>> >    OK, but I still don't see what that has to do
>> >    with proxies. That sounds like something an
>> >    edge router would do before it admits diffserv
>> >    traffic into the diffserv core. Proxies are even
>> >    more clueless because they never see the actual
>> >    traffic patterns.
>> >
>> > 		Mike
>> > Hfæj)bz	b²Ô^>X¬¶ÆÞ-YZnÇ(sm§ÿåSËlmée*¦ìr?¿(tm)¨¥(tm)©ÿ-+-SwèþÈ©
>
>
>
>_______________________________________________
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>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip
>


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From sip-admin@lists.bell-labs.com  Wed May 24 23:22:06 2000
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hi, Jonathan,

Thanks for your fast reply.

Actually it can be shown as following:

The SIP original operation:

           (1)       (2)       (3)       (4)       (5)
(caller)UAC--->proxy1--->proxy2--->proxy3--->proxy4--->UAS(callee)

Different proxy provides different service, the services are provided
sequentialy, all the (1),(2),(3),(4),(5) are standard SIP INVITE messages.

And another scheme:
           proxy2  proxy3  proxy4   
             (2)\  (3)/    (4)/  
(caller)UAC---->(local)proxy1---->UAS(callee)
            (1)               (5)

Now the services of proxy2, proxy3, proxy4 can be provided parallelly. The
(1),(5) 
are still standard SIP INVITE messages. But the (2),(3),(4) are simplified
requests which
only include the service provided by individual proxy, so (2),(3),(4)
messages are different 
and only part of original INVITE. Proxy1 sends out 3 requests
simultaneously, after it gets
the 3 reponses from 3 proxies, it rewrite the original INVITE and sends it
to UAS. 

By primarily consideration, the scheme can reduce call setup time, but is it
possible in SIP?

thanks
Yu








-----Original Message-----
From: Jonathan Rosenberg
To: Ge Yu
Cc: sip@lists.bell-labs.com
Sent: 5/25/00 2:35 AM
Subject: Re: [SIP] RE: about proxy



Ge Yu wrote:
> 
> hi, all,
> 
> If a UAC sends out an INVITE, which will tranverse 4 proxy servers
until it
> reaches the callee, the servers provide different services, this is
standard
> method of SIP.
> 
> There is another approach: UAC sends out an INVITE to the first proxy
> server, which may be its local proxy, then the proxy acts as a client
and
> sends the different service requests to the other 3 proxy servers.
Upon the
> first server receives the responses, it forwards the INVITE directly
to the
> callee, so the INVITE only need to tranverse ONE proxy.

I think you are talking about forking, but I'm not certain. SIP allows a
proxy to receive one request, and forward more than one request out. IN
this case, the local proxy would fork and send three INVITEs. These
would hit three separate servers, each of which tries to contact the
user. As they receive responses, these responses are forwarded upstream
to the forking proxy. The first call acceptance that the forking proxy
receives is forwarded upstream towards the caller.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Thu May 25 04:14:00 2000
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Ge Yu wrote:

> hi, Jonathan,
>
> Thanks for your fast reply.
>
> Actually it can be shown as following:
>
> The SIP original operation:
>
>            (1)       (2)       (3)       (4)       (5)
> (caller)UAC--->proxy1--->proxy2--->proxy3--->proxy4--->UAS(callee)
>
> Different proxy provides different service, the services are provided
> sequentialy, all the (1),(2),(3),(4),(5) are standard SIP INVITE messages.
>
> And another scheme:
>            proxy2  proxy3  proxy4
>              (2)\  (3)/    (4)/
> (caller)UAC---->(local)proxy1---->UAS(callee)
>             (1)               (5)
>
> Now the services of proxy2, proxy3, proxy4 can be provided parallelly. The
> (1),(5)
> are still standard SIP INVITE messages. But the (2),(3),(4) are simplified
> requests which
> only include the service provided by individual proxy, so (2),(3),(4)
> messages are different
> and only part of original INVITE. Proxy1 sends out 3 requests
> simultaneously, after it gets
> the 3 reponses from 3 proxies, it rewrite the original INVITE and sends it
> to UAS.

In the first example the service being performed by the SIP proxies is trying to
locate the callee, by routing the INVITE on to the next appropriate hop.

What are the services of P2, P3, P4 in your second example?

Whatever they are I think describing nodes 2,3,4  as SIP Proxy Servers is
leading to confusion. However, if they are SIP nodes offering services, that
fall within the solution space of SIP, it might be possible to add a new SIP
request method to do what you are describing.

hth,
Neil.
--
Ubiquity Software Corporation, UK           http://www.ubiquity.net



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From sip-admin@lists.bell-labs.com  Thu May 25 05:45:40 2000
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Date: Thu, 25 May 2000 11:44:28 +0200
From: Alberto Conte <Alberto.Conte@ms.alcatel.fr>
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Organization: Alcatel - Corporate Research Center
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Subject: [SIP] INVITE with incomplete SDP description
References: <75C79E507864D3118AFC00805FEAB7D85B272B@ripexch001.mcit.com>
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All,

I would like to allow a user to send an INVITE message carrying an incomplete session
description specifying the list of media types to be used in the session without
indicating the RTP ports and addresses for receiving them. 

That should be useful in case of third-party control, or gateway between two
signalling protocols.

I thought to something like this:

v=0
o=Alberto 2890844730 2890844730 IN IP4 100.101.102.103
s=Hello!
t=0 0
m=audio 0 RTP/AVP 3
a=rtpmap:3 GSM/8000
m=video 0 RTP/AVP 32
a=rtpmap:32 MPV/90000

i.e. putting the port number to zero. 

Is it possible? Can be detailed in the next RFC?

Thanks in advance fo the help,
Alberto


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From sip-admin@lists.bell-labs.com  Thu May 25 06:16:04 2000
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Subject: Re: [SIP] INVITE with incomplete SDP description
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Alberto Conte wrote:

> All,
>
> I would like to allow a user to send an INVITE message carrying an incomplete session
> description specifying the list of media types to be used in the session without
> indicating the RTP ports and addresses for receiving them.
>
> That should be useful in case of third-party control, or gateway between two
> signalling protocols.
>
> I thought to something like this:
>
> v=0
> o=Alberto 2890844730 2890844730 IN IP4 100.101.102.103
> s=Hello!
> t=0 0
> m=audio 0 RTP/AVP 3
> a=rtpmap:3 GSM/8000
> m=video 0 RTP/AVP 32
> a=rtpmap:32 MPV/90000
>
> i.e. putting the port number to zero.
>
> Is it possible? Can be detailed in the next RFC?

It already is used to infer a very different meaning from a callee to a caller. Look at
Appendix B - Usage of SDP ...

"If the callee wants neither to send nor receive a stream offered by the caller, the
callee sets the port number of that stream to zero in its media stream."

Cheers,
Neil.
--
Ubiquity Software Corporation, UK           http://www.ubiquity.net



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From sip-admin@lists.bell-labs.com  Thu May 25 06:38:59 2000
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From: Ge Yu <engp9374@nus.edu.sg>
To: Ge Yu <engp9374@nus.edu.sg>, "'Neil Deason'" <ndeason@ubiquity.net>
Cc: "'sip@lists.bell-labs.com '" <sip@lists.bell-labs.com>
Subject: RE: [SIP] RE: about proxy
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thanks, Neil,

1. About locate the callee, can we configure the location service as
follows:
   Maintain two kinds of location servers: global location server(should be
a series of servers, like DNS servers), which can provide the query service
to every caller;  local location server, which is maintained by local proxy
server.
    Before the callee moves to another place, he registers with his local
proxy, if his new place is in the same area of a specific local location
server, the proxy only updates the database in the local location server in
its domain.
Otherwise, if the new place is out of the range of the local location
server, the proxy requests to update the database in global location server.

    So if the callee moves frequently in a small area(that's quite possible
in reality), no need to update the global location server from time to time,
that is the usage of local location server.

2. If a caller wants to make a call, its local proxy only needs to query the
global location server, then it can get the latest location of the callee,
no need to tranverse many proxies to locate the callee.

So in the second example, the P2, P3,P4 may be location server, AAA server,
or other server which can provide subscribed services.

How do you think about this scheme?

Thanks for all reply.

Yu



> ----------
> From: 	Neil Deason[SMTP:ndeason@ubiquity.net]
> Sent: 	Thursday, May 25, 2000 4:11 PM
> To: 	Ge Yu
> Cc: 	'sip@lists.bell-labs.com '
> Subject: 	Re: [SIP] RE: about proxy
> 
> Ge Yu wrote:
> 
> > hi, Jonathan,
> >
> > Thanks for your fast reply.
> >
> > Actually it can be shown as following:
> >
> > The SIP original operation:
> >
> >            (1)       (2)       (3)       (4)       (5)
> > (caller)UAC--->proxy1--->proxy2--->proxy3--->proxy4--->UAS(callee)
> >
> > Different proxy provides different service, the services are provided
> > sequentialy, all the (1),(2),(3),(4),(5) are standard SIP INVITE
> messages.
> >
> > And another scheme:
> >            proxy2  proxy3  proxy4
> >              (2)\  (3)/    (4)/
> > (caller)UAC---->(local)proxy1---->UAS(callee)
> >             (1)               (5)
> >
> > Now the services of proxy2, proxy3, proxy4 can be provided parallelly.
> The
> > (1),(5)
> > are still standard SIP INVITE messages. But the (2),(3),(4) are
> simplified
> > requests which
> > only include the service provided by individual proxy, so (2),(3),(4)
> > messages are different
> > and only part of original INVITE. Proxy1 sends out 3 requests
> > simultaneously, after it gets
> > the 3 reponses from 3 proxies, it rewrite the original INVITE and sends
> it
> > to UAS.
> 
> In the first example the service being performed by the SIP proxies is
> trying to
> locate the callee, by routing the INVITE on to the next appropriate hop.
> 
> What are the services of P2, P3, P4 in your second example?
> 
> Whatever they are I think describing nodes 2,3,4  as SIP Proxy Servers is
> leading to confusion. However, if they are SIP nodes offering services,
> that
> fall within the solution space of SIP, it might be possible to add a new
> SIP
> request method to do what you are describing.
> 
> hth,
> Neil.
> --
> Ubiquity Software Corporation, UK           http://www.ubiquity.net
> 
> 


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From: "Holland, Peter Michael (Peter)" <pmholland@lucent.com>
To: sip@lists.bell-labs.com, "'Schroeder, Tim'" <schroeder@tri.sbc.com>
Subject: RE: [SIP] Carrying Q.BICC as an Mime-type in SIP
Date: Thu, 25 May 2000 11:30:21 +0100
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Tim,

> I agree that the information of type 1 is not relevant.  I'm just a little
> uncomfortable having the gateway on the BICC-to-SIP side edit out part of
> the BICC message to keep the SIP network happy.  It may edit out part of
> the
> BICC message for its own reasons (hiding the details of the source
> network?)
> but it shouldn't have to do so just because the SIP network can't
> encapsulate BICC, but only straight ISUP.  
> 
> I don't know of any reason why it *couldn't* edit it out, it just seems
> easier to pass along the whole BICC message transparently.  The
> destination
> BICC network can choose to use the parts it thinks are useful, and to
> ignore
> the parts it thinks aren't.  
> 
The normal BICC processing at the receiving side of the BICC-to-SIP gateway
will process and remove the type 1 information, because it is inherently
link-by-link according to the BICC procedures.

> In the ISUP case, we send the ISUP message transparently, without editing
> out the parts known (or suspected) to be useless in the destination
> network.
> I like that model, and it seems easy enough to follow for BICC as well.
> And
> even if BICC looks a lot like ISUP, and after the useless information has
> been removed may look exactly like ISUP, it still seems cleaner to
> transmit
> the BICC as encapsulated BICC, and label it as such with
> "application/bicc"
> or some such, rather than calling it ISUP. 
> 
I do not disagree with this in principle, but I do see a problem with how to
indicate this with the Content-Type as currently in
draft-ietf-sip-isup-mime-01.
Considering that:
1. a SIP-to-ISUP gateway ought to be able to handle a mime'd message that
originated from a previous BICC call segment, without being upgraded to
understand a new "BICC" Content-Type. (so it should be "application/ISUP")
2. there will be regional and national variants of BICC, just as there are
of ISUP.
(so the version, and base parameters are still needed as defined.)

The only reason I can see why a SIP-to-BICC gateway would be interested that
a previous leg was BICC is if there is some Type 2 information included. It
can
easily detect that by looking in the message, so maybe this is a sufficient
way of detecting this case.
If this is not considered sufficient then an additional parameter should be 
added to the Content-Type - "BICC=y/n"

> > The concern expressed refers to "converting to the ISUP format".  I'm
> not
> sure
> > what the concern really is, considering that the BICC message format is
> the same
> > as the ISUP message format, from the message type octet onwards.
> 
> Annex E of the Q.1901 document is all about BICC-to-ISUP interworking.
> The
> differences don't seem substantial, but ... 
> 
The differences are not substantial.

> If the BICC information is stripped of its BICC-ness and encapsulated as
> ISUP, will the destination end even be able to tell that it used to be
> BICC?
> Could that ever be useful?
> 
> Tim Schroeder
> tschroeder@tri.sbc.com
> SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759
> 
> 
> 
    Pete  Holland  
     Lucent Technologies,  Malmesbury, UK
     email: pmholland@lucent.com
     phone:  +44 1666 832423
     fax:  +44 1666 824515


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Date: Thu, 25 May 2000 13:01:37 +0200 (CEST)
From: Lars Berggren <lars@intertex.se>
To: Alberto Conte <Alberto.Conte@ms.alcatel.fr>
Cc: IETF SIP <sip@lists.bell-labs.com>
Subject: Re: [SIP] INVITE with incomplete SDP description
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On Thu, 25 May 2000, Alberto Conte wrote:

> All,
> 
> I would like to allow a user to send an INVITE message carrying an incomplete session
> description specifying the list of media types to be used in the session without
> indicating the RTP ports and addresses for receiving them. 
> 

If your intention is to only send streams, not to receive them, you can
list them as sendonly (a=sendonly). See section B.2 in rfc2543 and
rfc2327. /Lars

> That should be useful in case of third-party control, or gateway between two
> signalling protocols.
> 
> I thought to something like this:
> 
> v=0
> o=Alberto 2890844730 2890844730 IN IP4 100.101.102.103
> s=Hello!
> t=0 0
> m=audio 0 RTP/AVP 3
> a=rtpmap:3 GSM/8000
> m=video 0 RTP/AVP 32
> a=rtpmap:32 MPV/90000
> 
> i.e. putting the port number to zero. 
> 
> Is it possible? Can be detailed in the next RFC?
> 
> Thanks in advance fo the help,
> Alberto
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

Lars Berggren       <mailto:lars.berggren@intertex.se>
Intertex Data AB    tel: +46-8-6282828
Sundbyberg, Sweden  fax: +46-8-6286414



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Date: Thu, 25 May 2000 14:02:24 +0200
From: Alberto Conte <Alberto.Conte@ms.alcatel.fr>
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To: Neil Deason <ndeason@ubiquity.net>
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Subject: Re: [SIP] INVITE with incomplete SDP description
References: <75C79E507864D3118AFC00805FEAB7D85B272B@ripexch001.mcit.com> <392CF5FC.E49FAF7F@ms.alcatel.fr> <392CFCB8.3B21640@ubiquity.net>
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Neil Deason a écrit :
> 
> Alberto Conte wrote:
> 
> > All,
> >
> > I would like to allow a user to send an INVITE message carrying an incomplete session
> > description specifying the list of media types to be used in the session without
> > indicating the RTP ports and addresses for receiving them.
> >
> > That should be useful in case of third-party control, or gateway between two
> > signalling protocols.
> >
> > I thought to something like this:
> >
> > v=0
> > o=Alberto 2890844730 2890844730 IN IP4 100.101.102.103
> > s=Hello!
> > t=0 0
> > m=audio 0 RTP/AVP 3
> > a=rtpmap:3 GSM/8000
> > m=video 0 RTP/AVP 32
> > a=rtpmap:32 MPV/90000
> >
> > i.e. putting the port number to zero.
> >
> > Is it possible? Can be detailed in the next RFC?
> 
> It already is used to infer a very different meaning from a callee to a caller. Look at
> Appendix B - Usage of SDP ...
> 
> "If the callee wants neither to send nor receive a stream offered by the caller, the
> callee sets the port number of that stream to zero in its media stream."
> 
> Cheers,
> Neil.

Thank you Neil for your answer. 

I know the content of Appendix B and I agree with you. However, as you say, the
"port=0" meaning has been fixed only in the case of calle -> caller response or in a
caller -> callee ACK, NOT in the caller -> callee INVITE message. 

I still think that is acceptable to state that the port=0 in an INVITE message means:
"I invite you for a session according with these medias, but at the moment I don't
have the RTP port and address". 

RTP port and address will be sent in the ACK!

Note that in my case the caller is not asking for a send-only session, but only
delaying the port and address information.

Cheers,
Alberto

> --
> Ubiquity Software Corporation, UK           http://www.ubiquity.net
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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From sip-admin@lists.bell-labs.com  Thu May 25 08:39:44 2000
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From: Alberto Conte <Alberto.Conte@ms.alcatel.fr>
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Subject: Re: [SIP] INVITE with incomplete SDP description
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Lars Berggren a écrit :
> 
> On Thu, 25 May 2000, Alberto Conte wrote:
> 
> > Neil Deason a écrit :
> > >
> > > Alberto Conte wrote:
> > >
> > > > All,
> > > >
> > > > I would like to allow a user to send an INVITE message carrying an incomplete session
> > > > description specifying the list of media types to be used in the session without
> > > > indicating the RTP ports and addresses for receiving them.
> > > >
> > > > That should be useful in case of third-party control, or gateway between two
> > > > signalling protocols.
> > > >
> > > > I thought to something like this:
> > > >
> > > > v=0
> > > > o=Alberto 2890844730 2890844730 IN IP4 100.101.102.103
> > > > s=Hello!
> > > > t=0 0
> > > > m=audio 0 RTP/AVP 3
> > > > a=rtpmap:3 GSM/8000
> > > > m=video 0 RTP/AVP 32
> > > > a=rtpmap:32 MPV/90000
> > > >
> > > > i.e. putting the port number to zero.
> > > >
> > > > Is it possible? Can be detailed in the next RFC?
> > >
> > > It already is used to infer a very different meaning from a callee to a caller. Look at
> > > Appendix B - Usage of SDP ...
> > >
> > > "If the callee wants neither to send nor receive a stream offered by the caller, the
> > > callee sets the port number of that stream to zero in its media stream."
> > >
> > > Cheers,
> > > Neil.
> >
> > Thank you Neil for your answer.
> >
> > I know the content of Appendix B and I agree with you. However, as you say, the
> > "port=0" meaning has been fixed only in the case of calle -> caller response or in a
> > caller -> callee ACK, NOT in the caller -> callee INVITE message.
> >
> > I still think that is acceptable to state that the port=0 in an INVITE message means:
> > "I invite you for a session according with these medias, but at the moment I don't
> > have the RTP port and address".
> >
> > RTP port and address will be sent in the ACK!
> >
> > Note that in my case the caller is not asking for a send-only session, but only
> > delaying the port and address information.
> 
> In this case you can send an INVITE without SDP, the callee lists streams
> in the 200, and the ACK can contain the final SDP. This behaviour is
> referenced in the spec. /Lars
> 

Sorry Lars, I don't agree with you.

Sending an INVITE without SDP means that the caller will be forced to choose the
medias exchanged in the session on the base of the ones proposed by the callee! 

This is not exactly what I want. I would like the caller able to force the callee to
accept its choice (or a part) of the medias, delaying only the RTP port and address
information.

> >
> > Cheers,
> > Alberto
> >
> > > --
> > > Ubiquity Software Corporation, UK           http://www.ubiquity.net
> > >
> > > _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > http://lists.bell-labs.com/mailman/listinfo/sip
> >
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> >
> 
> Lars Berggren       <mailto:lars.berggren@intertex.se>
> Intertex Data AB    tel: +46-8-6282828
> Sundbyberg, Sweden  fax: +46-8-6286414


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From sip-admin@lists.bell-labs.com  Thu May 25 09:01:22 2000
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Subject: Re: [SIP] INVITE with incomplete SDP description
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Alberto Conte wrote:

> Lars Berggren a écrit :
> >

<snip>

> > In this case you can send an INVITE without SDP, the callee lists streams
> > in the 200, and the ACK can contain the final SDP. This behaviour is
> > referenced in the spec. /Lars
> >
>
> Sorry Lars, I don't agree with you.
>
> Sending an INVITE without SDP means that the caller will be forced to choose the
> medias exchanged in the session on the base of the ones proposed by the callee!
>
> This is not exactly what I want. I would like the caller able to force the callee to
> accept its choice (or a part) of the medias, delaying only the RTP port and address
> information.

But by specifying the medias in an INVITE you don't 'force' the callee to accept the caller's
choice. It is the opening offer in the two parties agreeing on media types. If you do not include
the session description in the INVITE, the other party should offer it's supported media streams
and codecs in its 1xx or 2xx response. Your UA then sends a completed session description in the
ACK.

Cheers,
Neil.
--
Ubiquity Software Corporation, UK           http://www.ubiquity.net




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From sip-admin@lists.bell-labs.com  Thu May 25 09:58:37 2000
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Date: Thu, 25 May 2000 09:58:17 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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Based on earlier separate threads, we currently have both
Content-Function and Content-Disposition in the 2543bis draft. They seem
to overlap to some large extent, so a decision needs to be reached as to
whether they are truly different things or just one and, if one, which
of the two to pick. Content-Disposition is an existing concept (RFC
2183), while Content-Function is not used outside of SIP. My preference
would be to stick with Content-Disposition.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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Date: Thu, 25 May 2000 14:10:04 +0200 (CEST)
From: Lars Berggren <lars@intertex.se>
To: Alberto Conte <Alberto.Conte@ms.alcatel.fr>
Cc: Neil Deason <ndeason@ubiquity.net>, IETF SIP <sip@lists.bell-labs.com>
Subject: Re: [SIP] INVITE with incomplete SDP description
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On Thu, 25 May 2000, Alberto Conte wrote:

> Neil Deason a écrit :
> > 
> > Alberto Conte wrote:
> > 
> > > All,
> > >
> > > I would like to allow a user to send an INVITE message carrying an incomplete session
> > > description specifying the list of media types to be used in the session without
> > > indicating the RTP ports and addresses for receiving them.
> > >
> > > That should be useful in case of third-party control, or gateway between two
> > > signalling protocols.
> > >
> > > I thought to something like this:
> > >
> > > v=0
> > > o=Alberto 2890844730 2890844730 IN IP4 100.101.102.103
> > > s=Hello!
> > > t=0 0
> > > m=audio 0 RTP/AVP 3
> > > a=rtpmap:3 GSM/8000
> > > m=video 0 RTP/AVP 32
> > > a=rtpmap:32 MPV/90000
> > >
> > > i.e. putting the port number to zero.
> > >
> > > Is it possible? Can be detailed in the next RFC?
> > 
> > It already is used to infer a very different meaning from a callee to a caller. Look at
> > Appendix B - Usage of SDP ...
> > 
> > "If the callee wants neither to send nor receive a stream offered by the caller, the
> > callee sets the port number of that stream to zero in its media stream."
> > 
> > Cheers,
> > Neil.
> 
> Thank you Neil for your answer. 
> 
> I know the content of Appendix B and I agree with you. However, as you say, the
> "port=0" meaning has been fixed only in the case of calle -> caller response or in a
> caller -> callee ACK, NOT in the caller -> callee INVITE message. 
> 
> I still think that is acceptable to state that the port=0 in an INVITE message means:
> "I invite you for a session according with these medias, but at the moment I don't
> have the RTP port and address". 
> 
> RTP port and address will be sent in the ACK!
> 
> Note that in my case the caller is not asking for a send-only session, but only
> delaying the port and address information.

In this case you can send an INVITE without SDP, the callee lists streams
in the 200, and the ACK can contain the final SDP. This behaviour is
referenced in the spec. /Lars

> 
> Cheers,
> Alberto
> 
> > --
> > Ubiquity Software Corporation, UK           http://www.ubiquity.net
> > 
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
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> 

Lars Berggren       <mailto:lars.berggren@intertex.se>
Intertex Data AB    tel: +46-8-6282828
Sundbyberg, Sweden  fax: +46-8-6286414



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From: "Sinnreich, Henry" <Henry.Sinnreich@WCOM.Com>
To: "'sip@lists.bell-labs.com '" <sip@lists.bell-labs.com>
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You may have seen it, but here is the URL:
http://commweb.com/features/comptel/2k0522.sip.html

Henry


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From: "Mark Watson" <mwatson@nortelnetworks.com>
To: "'Schroeder, Tim'" <schroeder@tri.sbc.com>,
        "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>
Subject: RE: [SIP] Carrying Q.BICC as an Mime-type in SIP
Date: Thu, 25 May 2000 13:19:54 +0100
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Tim,

If we followed your proposal, then there would be no way for CS(C) to
distibguish between case (i) and case (ii). In both cases, CS(C) would
receive a SIP INVITE message containing a BICC IAM (itself containing Bearer
Control information) and also Bearer COntrol information within the SDP.

In case (i), the BICC APM information will relate to the MG controlled by
CS(A) (MG(A)), as will the SDP.

In case (ii), the BICC APM information will related to MG(A) and the SDP
will relate to MG(B).

The only sensible behaviour for CS(C) is to use the SDP in both cases, so
the BICC information will never be used, even in the BICC -- SIP -- BICC
case.

The BICC APM is essentially 'link by link' between Call Servers. It is
required that CS(B) can interpret this information and so implicitly it has
the capability to remove it, turning the BICC message back into an ISUP one.

It will be easier to come to a clear conclusion on this one when we have a
better idea of how BICC will support IP. One option being discussed is that
BICC will support carriage of SDP information, which will be 'tunnelled'
transparently from one MG, via the Call Servers, to the next MG. This
information would be rather easy to interwork to SIP.

Regards,

Mark Watson
Nortel Networks

-----Original Message-----
From: Schroeder, Tim [mailto:schroeder@tri.sbc.com]
Sent: 23 May 2000 22:25
To: sip@lists.bell-labs.com
Subject: RE: [SIP] Carrying Q.BICC as an Mime-type in SIP


Mark Watson [mailto:mwatson@nortelnetworks.com] writes:

 If we have CS(A) --bicc ---CS(B)---sip --- CS(C)---bicc---CS(D) 

where 'CS' is 'Call Server', for want of a better term, then there are two
cases: 

(i) with the bearer path routed directly between Gateways controlled by
CS(A) and CS(D), and 

(ii) with further 'packet-to-packet' gateways contolled by CS(B) and CS(C)
i.e. there are three 'domains' which are physically interworked at both the
call and bearer layer.

There is some doubt as to whether (i) will be possible at all - this depends
on how the support of IP in BICC is defined over the next few months. If it
is possible, I think that it should involve maping the bearer control
mechanism in BICC to the equivalent in SIP (SDP) so that the call can still
work if it terminates on a pure SIP agent. 

Agreed.

If you do this mapping, then you've removed from the BICC information
everything which makes it BICC and not ISUP, so you're not carrying BICC
inside SIP at all. 

But just because you map the information from BICC into SIP, doesn't mean
it's removed from the BICC message, does it?  In the ISUP--SIP--ISUP case,
the ISUP information that makes sense to map to SIP (to provide the desired
level of interworking) is mapped, but the ISUP message is still passed along
in the body to allow for ISUP transparency, in case the call ends up back on
an ISUP-signaled network.  I'd think the same mechanism would apply to the
BICC--SIP--BICC case: map the BICC information that makes sense to SIP (and
the bearer stuff to SDP), but pass along the original BICC message in the
body.  Then if the call terminates in the SIP-signaled network, you've got
your interworking (to the level that you coherently mapped the BICC
information to SIP/SDP), and if the call is routed through a gateway back to
a BICC-signaled network you can still have "BICC transparency".

In case (ii), the three domains are completely independent from each other
when it comes to setup of the bearer connection. Again, the only information
from BICC left to carry over the SIP domain is the ISUP part. 

I think I'd still want to send the whole BICC message, even if the non-ISUP
part will never be used.  The gateways provide the interworking
functionality between BICC and SIP; there's no need to have them also
interwork between BICC and ISUP (even if that's relatively easy) just so I
can avoid an application/BICC or application/Q.1901 MIME type. 

If we were to carry the BICC APM parameter, which contains the Bearer
related information, over SIP, we would be inventing a new way of carrying
out bearer setup on a SIP call - an alternative to SDP. I do not think we
should do this as it would be non-backwards-compatible. 

In your case (i), interwork the bearer related information to SDP (which
will get used if the call terminates in the SIP-signaled network, ignoring
the BICC-embedded APM parameter) and leave it in the BICC message as well,
passing the whole BICC message along in the body (which will get used if the
call is routed back to the BICC-signaled network).  

In your case (ii), the gateways are going to have to piece together the
three media streams end-to-end, a first BICC APM parameter will describe the
CS(A)--CS(B) segment, SDP will describe the CS(B)--CS(C) segment, and a
second BICC APM parameter will describe the CS(C)--CS(D) segment.  I don't
think there's much BICC/APM-to-SDP interworking that can be done there, but
the BICC message may as well be passed along, rather than first converting
it to the ISUP format.

Tim Schroeder 
tschroeder@tri.sbc.com 
SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759 

 


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2651.65">
<TITLE>RE: [SIP] Carrying Q.BICC as an Mime-type in SIP</TITLE>
</HEAD>
<BODY>

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

<P><FONT SIZE=3D2>If we followed your proposal, then there would be no =
way for CS(C) to distibguish between case (i) and case (ii). In both =
cases, CS(C) would receive a SIP INVITE message containing a BICC IAM =
(itself containing Bearer Control information) and also Bearer COntrol =
information within the SDP.</FONT></P>

<P><FONT SIZE=3D2>In case (i), the BICC APM information will relate to =
the MG controlled by CS(A) (MG(A)), as will the SDP.</FONT>
</P>

<P><FONT SIZE=3D2>In case (ii), the BICC APM information will related =
to MG(A) and the SDP will relate to MG(B).</FONT>
</P>

<P><FONT SIZE=3D2>The only sensible behaviour for CS(C) is to use the =
SDP in both cases, so the BICC information will never be used, even in =
the BICC -- SIP -- BICC case.</FONT></P>

<P><FONT SIZE=3D2>The BICC APM is essentially 'link by link' between =
Call Servers. It is required that CS(B) can interpret this information =
and so implicitly it has the capability to remove it, turning the BICC =
message back into an ISUP one.</FONT></P>

<P><FONT SIZE=3D2>It will be easier to come to a clear conclusion on =
this one when we have a better idea of how BICC will support IP. One =
option being discussed is that BICC will support carriage of SDP =
information, which will be 'tunnelled' transparently from one MG, via =
the Call Servers, to the next MG. This information would be rather easy =
to interwork to SIP.</FONT></P>

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

<P><FONT SIZE=3D2>Mark Watson</FONT>
<BR><FONT SIZE=3D2>Nortel Networks</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Schroeder, Tim [<A =
HREF=3D"mailto:schroeder@tri.sbc.com">mailto:schroeder@tri.sbc.com</A>]<=
/FONT>
<BR><FONT SIZE=3D2>Sent: 23 May 2000 22:25</FONT>
<BR><FONT SIZE=3D2>To: sip@lists.bell-labs.com</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [SIP] Carrying Q.BICC as an Mime-type =
in SIP</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Mark Watson [<A =
HREF=3D"mailto:mwatson@nortelnetworks.com">mailto:mwatson@nortelnetworks=
.com</A>] writes:</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;If we have CS(A) --bicc ---CS(B)---sip --- =
CS(C)---bicc---CS(D) </FONT>
</P>

<P><FONT SIZE=3D2>where 'CS' is 'Call Server', for want of a better =
term, then there are two</FONT>
<BR><FONT SIZE=3D2>cases: </FONT>
</P>

<P><FONT SIZE=3D2>(i) with the bearer path routed directly between =
Gateways controlled by</FONT>
<BR><FONT SIZE=3D2>CS(A) and CS(D), and </FONT>
</P>

<P><FONT SIZE=3D2>(ii) with further 'packet-to-packet' gateways =
contolled by CS(B) and CS(C)</FONT>
<BR><FONT SIZE=3D2>i.e. there are three 'domains' which are physically =
interworked at both the</FONT>
<BR><FONT SIZE=3D2>call and bearer layer.</FONT>
</P>

<P><FONT SIZE=3D2>There is some doubt as to whether (i) will be =
possible at all - this depends</FONT>
<BR><FONT SIZE=3D2>on how the support of IP in BICC is defined over the =
next few months. If it</FONT>
<BR><FONT SIZE=3D2>is possible, I think that it should involve maping =
the bearer control</FONT>
<BR><FONT SIZE=3D2>mechanism in BICC to the equivalent in SIP (SDP) so =
that the call can still</FONT>
<BR><FONT SIZE=3D2>work if it terminates on a pure SIP agent. </FONT>
</P>

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

<P><FONT SIZE=3D2>If you do this mapping, then you've removed from the =
BICC information</FONT>
<BR><FONT SIZE=3D2>everything which makes it BICC and not ISUP, so =
you're not carrying BICC</FONT>
<BR><FONT SIZE=3D2>inside SIP at all. </FONT>
</P>

<P><FONT SIZE=3D2>But just because you map the information from BICC =
into SIP, doesn't mean</FONT>
<BR><FONT SIZE=3D2>it's removed from the BICC message, does it?&nbsp; =
In the ISUP--SIP--ISUP case,</FONT>
<BR><FONT SIZE=3D2>the ISUP information that makes sense to map to SIP =
(to provide the desired</FONT>
<BR><FONT SIZE=3D2>level of interworking) is mapped, but the ISUP =
message is still passed along</FONT>
<BR><FONT SIZE=3D2>in the body to allow for ISUP transparency, in case =
the call ends up back on</FONT>
<BR><FONT SIZE=3D2>an ISUP-signaled network.&nbsp; I'd think the same =
mechanism would apply to the</FONT>
<BR><FONT SIZE=3D2>BICC--SIP--BICC case: map the BICC information that =
makes sense to SIP (and</FONT>
<BR><FONT SIZE=3D2>the bearer stuff to SDP), but pass along the =
original BICC message in the</FONT>
<BR><FONT SIZE=3D2>body.&nbsp; Then if the call terminates in the =
SIP-signaled network, you've got</FONT>
<BR><FONT SIZE=3D2>your interworking (to the level that you coherently =
mapped the BICC</FONT>
<BR><FONT SIZE=3D2>information to SIP/SDP), and if the call is routed =
through a gateway back to</FONT>
<BR><FONT SIZE=3D2>a BICC-signaled network you can still have =
&quot;BICC transparency&quot;.</FONT>
</P>

<P><FONT SIZE=3D2>In case (ii), the three domains are completely =
independent from each other</FONT>
<BR><FONT SIZE=3D2>when it comes to setup of the bearer connection. =
Again, the only information</FONT>
<BR><FONT SIZE=3D2>from BICC left to carry over the SIP domain is the =
ISUP part. </FONT>
</P>

<P><FONT SIZE=3D2>I think I'd still want to send the whole BICC =
message, even if the non-ISUP</FONT>
<BR><FONT SIZE=3D2>part will never be used.&nbsp; The gateways provide =
the interworking</FONT>
<BR><FONT SIZE=3D2>functionality between BICC and SIP; there's no need =
to have them also</FONT>
<BR><FONT SIZE=3D2>interwork between BICC and ISUP (even if that's =
relatively easy) just so I</FONT>
<BR><FONT SIZE=3D2>can avoid an application/BICC or application/Q.1901 =
MIME type. </FONT>
</P>

<P><FONT SIZE=3D2>If we were to carry the BICC APM parameter, which =
contains the Bearer</FONT>
<BR><FONT SIZE=3D2>related information, over SIP, we would be inventing =
a new way of carrying</FONT>
<BR><FONT SIZE=3D2>out bearer setup on a SIP call - an alternative to =
SDP. I do not think we</FONT>
<BR><FONT SIZE=3D2>should do this as it would be =
non-backwards-compatible. </FONT>
</P>

<P><FONT SIZE=3D2>In your case (i), interwork the bearer related =
information to SDP (which</FONT>
<BR><FONT SIZE=3D2>will get used if the call terminates in the =
SIP-signaled network, ignoring</FONT>
<BR><FONT SIZE=3D2>the BICC-embedded APM parameter) and leave it in the =
BICC message as well,</FONT>
<BR><FONT SIZE=3D2>passing the whole BICC message along in the body =
(which will get used if the</FONT>
<BR><FONT SIZE=3D2>call is routed back to the BICC-signaled =
network).&nbsp; </FONT>
</P>

<P><FONT SIZE=3D2>In your case (ii), the gateways are going to have to =
piece together the</FONT>
<BR><FONT SIZE=3D2>three media streams end-to-end, a first BICC APM =
parameter will describe the</FONT>
<BR><FONT SIZE=3D2>CS(A)--CS(B) segment, SDP will describe the =
CS(B)--CS(C) segment, and a</FONT>
<BR><FONT SIZE=3D2>second BICC APM parameter will describe the =
CS(C)--CS(D) segment.&nbsp; I don't</FONT>
<BR><FONT SIZE=3D2>think there's much BICC/APM-to-SDP interworking that =
can be done there, but</FONT>
<BR><FONT SIZE=3D2>the BICC message may as well be passed along, rather =
than first converting</FONT>
<BR><FONT SIZE=3D2>it to the ISUP format.</FONT>
</P>

<P><FONT SIZE=3D2>Tim Schroeder </FONT>
<BR><FONT SIZE=3D2>tschroeder@tri.sbc.com </FONT>
<BR><FONT SIZE=3D2>SBC Technology Resources, 9505 Arboretum Blvd., =
Austin, TX&nbsp; 78759 </FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;</FONT>
</P>
<BR>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>SIP mailing list</FONT>
<BR><FONT SIZE=3D2>SIP@lists.bell-labs.com</FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://lists.bell-labs.com/mailman/listinfo/sip" =
TARGET=3D"_blank">http://lists.bell-labs.com/mailman/listinfo/sip</A></F=
ONT>
</P>

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Let me add that that story was the cover article of the May Computer
Telephony Magazine. The cover itself is very amusing.... I encourage
people to pick up a copy.

-Jonathan R.

"Sinnreich, Henry" wrote:
> 
> You may have seen it, but here is the URL:
> http://commweb.com/features/comptel/2k0522.sip.html
> 
> Henry
> 
> >SIP mailing list
> >SIP@lists.bell-labs.com
> >http://lists.bell-labs.com/mailman/listinfo/sip
> >
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Ping doesn't really tell me much about load distribution between traffic
priorities, nor about timing variations. Maybe a priority PING . . .

--
Dean

Dave Oran wrote:
> 
> > Or maybe we need a general "Hey, how's the net doing?" protocol . . .
> >
> Ping?


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Currently, Contact is optional for OPTIONS. While this is harmless, it
is not quite clear what this Contact would mean in this context. It
could mean "send future requests here", if used within an existing call.
Any preferences?
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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From sip-admin@lists.bell-labs.com  Thu May 25 11:06:47 2000
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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        IETF SIP <sip@lists.bell-labs.com>
Subject: Re: [SIP] INVITE with incomplete SDP description
References: <Pine.LNX.4.02.10005251407060.1740-100000@lars.intertex.se> <392D1D45.5259F3DF@ms.alcatel.fr> <392D2397.EA6AFAC8@ubiquity.net>
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Another way to do it is to send the initial INVITE with the desired
media streams on hold (IP address set to 0.0.0.0). When you have
specific IP address and port information, you can do a re-INVITE, and
take the user off hold by passing an updated IP address and port in each
m line. Note that the update would not occur in an ACK, but rather in a
re-INVITE.

-Jonathan R.



Neil Deason wrote:
> 
> Alberto Conte wrote:
> 
> > Lars Berggren a écrit :
> > >
> 
> <snip>
> 
> > > In this case you can send an INVITE without SDP, the callee lists streams
> > > in the 200, and the ACK can contain the final SDP. This behaviour is
> > > referenced in the spec. /Lars
> > >
> >
> > Sorry Lars, I don't agree with you.
> >
> > Sending an INVITE without SDP means that the caller will be forced to choose the
> > medias exchanged in the session on the base of the ones proposed by the callee!
> >
> > This is not exactly what I want. I would like the caller able to force the callee to
> > accept its choice (or a part) of the medias, delaying only the RTP port and address
> > information.
> 
> But by specifying the medias in an INVITE you don't 'force' the callee to accept the caller's
> choice. It is the opening offer in the two parties agreeing on media types. If you do not include
> the session description in the INVITE, the other party should offer it's supported media streams
> and codecs in its 1xx or 2xx response. Your UA then sends a completed session description in the
> ACK.
> 
> Cheers,
> Neil.
> --
> Ubiquity Software Corporation, UK           http://www.ubiquity.net
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Thu May 25 11:23:21 2000
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Subject: Re: [SIP] Contact in OPTIONS
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To date, OPTIONS has been one of those "outside of the call" methods
which has no impact on call state. Changing the address for routing
subsequent requests was something I thought was handled through
re-INVITE; having this also happen through OPTIONS seems to add another
thing to check for, and also means that OPTIONS now has an effect on
calls, when it didn't before.

A reasonable use for Contact in OPTIONS (and its response) is to list
the communications means available for establishing new sessions or
other things. For example:

OPTIONS sip:hgs@cs.columbia.edu SIP/2.0
From: sip:jdrosen@dynamicsoft.com
Contact: sip:jdrosen@dynamicsoft.com, http://www.cs.columbia.edu
 ,mailto:jdrosen@dynamicsoft.com

and the reply contains Contact headers with the same thing.

-Jonathan .

Henning Schulzrinne wrote:
> 
> Currently, Contact is optional for OPTIONS. While this is harmless, it
> is not quite clear what this Contact would mean in this context. It
> could mean "send future requests here", if used within an existing call.
> Any preferences?
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From: Alberto Conte <Alberto.Conte@ms.alcatel.fr>
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        IETF SIP <sip@lists.bell-labs.com>
Subject: Re: [SIP] INVITE with incomplete SDP description
References: <Pine.LNX.4.02.10005251407060.1740-100000@lars.intertex.se> <392D1D45.5259F3DF@ms.alcatel.fr> <392D2397.EA6AFAC8@ubiquity.net> <392D43E7.CC2F69C2@dynamicsoft.com>
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Putting in hold the calle is a solution, but that increases the number of messages to
be exchanged, requiring a re-INVITE transaction!

The power of SIP is to reduce the message exchanged in case of no error. And this is
actually a case of no-error. It should be better allowing the caller to propose its
media request delaying only the RTP port information in the ACK.

Alberto 

Jonathan Rosenberg wrote :
> 
> Another way to do it is to send the initial INVITE with the desired
> media streams on hold (IP address set to 0.0.0.0). When you have
> specific IP address and port information, you can do a re-INVITE, and
> take the user off hold by passing an updated IP address and port in each
> m line. Note that the update would not occur in an ACK, but rather in a
> re-INVITE.
> 
> -Jonathan R.
> 
> Neil Deason wrote:
> >
> > Alberto Conte wrote:
> >
> > > Lars Berggren a écrit :
> > > >
> >
> > <snip>
> >
> > > > In this case you can send an INVITE without SDP, the callee lists streams
> > > > in the 200, and the ACK can contain the final SDP. This behaviour is
> > > > referenced in the spec. /Lars
> > > >
> > >
> > > Sorry Lars, I don't agree with you.
> > >
> > > Sending an INVITE without SDP means that the caller will be forced to choose the
> > > medias exchanged in the session on the base of the ones proposed by the callee!
> > >
> > > This is not exactly what I want. I would like the caller able to force the callee to
> > > accept its choice (or a part) of the medias, delaying only the RTP port and address
> > > information.
> >
> > But by specifying the medias in an INVITE you don't 'force' the callee to accept the caller's
> > choice. It is the opening offer in the two parties agreeing on media types. If you do not include
> > the session description in the INVITE, the other party should offer it's supported media streams
> > and codecs in its 1xx or 2xx response. Your UA then sends a completed session description in the
> > ACK.
> >
> > Cheers,
> > Neil.
> > --
> > Ubiquity Software Corporation, UK           http://www.ubiquity.net
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com


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> -----Original Message-----
> From: sip-admin@lists.bell-labs.com
> [mailto:sip-admin@lists.bell-labs.com]On Behalf Of Jiri Kuthan
> Sent: Wednesday, May 24, 2000 6:13 PM
> To: Dave Oran
> Cc: Dean Willis; Michael Thomas; IETF SIP
> Subject: Re: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task
> force in Adelaide)
>
>
> Dave,
>
> In the DiffServ case, I see these policies to control the edge:
>
> 0) no control; the edge trusts end-devices absolutely; end-devices
>    will get what they want
> 	-> not very secure
It also provides no traffic isloation, so one badly behaving host can
degrade or deny service for a whole class. Call this a security problem if
you want, but I don't. Diffserv by itself doesn't do "admission control",
only aggregate policing.

> 1) self-control; the edge manages itself; it uses some heuristics to say
>    which packet deserves which class
> 	-> heuristics may fail
If you believe Diffserv BY ITSELF is adequate, then by definition the
heuristics don't fail. They only determine who wins and loses when the
policing starts remarking or tossing packets.

> 2) the edge is polled/controlled explicitly by
>    a) end-devices
> 	->they have some built-in control support, e.g. RSVP mapped to
>           DiffServe at the edge
Yes.

>    b) proxies on behalf of the end-devices
> 	->works w/o changes to end-devices
Define "works". I'm skeptical. This requires that the proxy knows enough
about the topology to know where the end-device hits the edge, and knows
enough about the current SLAs and policing parameters at that edge device to
know if allowing the call will exceed the SLA and trigger aggregate
policing.

> 	->application-awareness may be added on top of the
> 	  admission control, e.g. a SIP/RTSP/quake proxy will be
>           able to decide that all video streams from
> 	  head of department will be assigned high
> 	  priority
>
This is a completely independent issue. RSVP and COPS now have
priority/preemption objects which allow admission control to take this kind
of policy into account.

> The question: Is 2b an invalid policy?
I'm a registered skeptic. I don't think it has well defined properties such
that you can explain either to a customer or an end user why the system
behaves in a certain way. If you buy the notion that you don't need
resource-availability-based admission control and traffic isolation to
confine the effects of oversubscription, then I suppose 2b is workable. I
wouldn't buy it. Someone else might.

> Is it a
> useful policy? And what is actually the difference in terms
> of scalability between
> i) end-devices trying to control/poll the edge [2a]
> ii) farm of SIP proxies doing the same on behalf of them
>     (possibly using some kind of aggregation) [2b]?
> The scalability bottleneck is the edge in both cases,
Not really. The edge is in fact the only place where I think the scalability
problem is tractable, because that's the point where the policing is done.

> isn't it?
>
> Thanks,
>
> Jiri
>
> Dave Oran wrote:
> >
> > Ah. I understand. no RSVP == no explicit admission control.
> > That's the bug. Hijacking SIP proxies to do the job of a bandwidth
> > allocation and admission control signaling protocol is ugly at
> best. You'd
> > need one of these beasts for every application in the universe - rtsp,
> > quake, you name it.
> >
> > If you build SIP proxies the way some people build web server
> farms you'll
> > have an entirely different scaling problem, which is 100's of
> SIP proxies
> > polling a poor overworked network managmeent system trying to figure out
> > what it knows about the state of the core (or better stated,
> what it knew
> > about the core 5 minutes ago when it last gathered enough
> measurements to
> > make a guess).
> >
> > Register me as a skeptic.
> >
> > Dave.
> >
> > > -----Original Message-----
> > > From: Dean Willis [mailto:dwillis@dynamicsoft.com]
> > > Sent: Wednesday, May 24, 2000 3:53 PM
> > > To: Dave Oran
> > > Cc: Dean Willis; Michael Thomas; IETF SIP
> > > Subject: Re: SIP DiffServe (was RE: [SIP] Minutes of SIP Security task
> > > force in Adelaide)
> > >
> > >
> > >
> > >
> > > Dave Oran wrote:
> > > >
> > > > If you're using Intserv at the edge the router at the edge of
> > > the Diffserv
> > > > cloud will likely know this and reject the reservation. No
> need for the
> > > > proxy to get involved.
> > >
> > > In general, I'm not. The WorldCom net, for example, is
> TOS-priority from
> > > end to end.
> > >
> > > >
> > > > The car analogy is not a good one because in the case of
> highways you're
> > > > already stuck at the bottleneck when you find out you're
> > > screwed, as opposed
> > > > to RSVP when you are essentailly clearing all the dead wrecks
> > > out of your
> > > > way before driving up the highway, ala Mad Max.
> > > >
> > > > Positing that a SIP proxy has a better idea what the QoS
> state of the
> > > > Diffserv core is than the edge router at the edge of that
> core is like
> > > > saying the radio announcer knows more about the tornado
> than the Twister
> > > > crew chasing it.
> > >
> > > A SIP proxy is in a better position to have a persistent relationship
> > > with a network management system which can feed it
> information about the
> > > state of the network than a dynamically romin SIP UA is. So,
> even though
> > > the announceer may not know as much about the tornado as the chaser
> > > does, he is in a MUCH better position to share that
> information with the
> > > listening audience.
> > >
> > > > If there's a oracle which knows the state of the core it can
> > > tell the edges
> > > > as easily or MORE easily than it can tell an undefined sea of
> > > application
> > > > proxy boxes.
> > >
> > > Sure, the edges probably know truckloads of cool stuff.
> > >
> > > But I have no mechanism for the edges to relay this knowledge to the
> > > UAs.
> > >
> > > --
> > > Dean
> > >
> > >
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
>
> --
> Jiri Kuthan               Phone: +49-30-3463 7271
> GMD FOKUS                 Fax  : +49-30-3463 8271
> Kaiserin-Augusta-Allee 31 E-Mail : kuthan@fokus.gmd.de
> D-10589 Berlin, Germany   WWW    : http://www.fokus.gmd.de/usr/kuthan
> --
> Curious about Internet Telephony?
> Check http://www.fokus.gmd.de/glone/projects/ipt today!
>
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>
>



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In Figure 4 in 8.3 Add Party during Add Party, D, acting on the untriggered INVITE
message from B, sends an INVITE message to A, "INV A Also:A, B."  But, shouldn't
this INVITE message from D be "INV A Also:B ReqBy:B" instead?

--Hyong




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They definitely should be consolidated. Since inline, at least, does
make sense for us, I would vote to use the HTTP version,
Content-Disposition. We definitely need to also have the
optional/required bit, which is critical for correct multipart usage.

-Jonathan R.

Henning Schulzrinne wrote:
> 
> Based on earlier separate threads, we currently have both
> Content-Function and Content-Disposition in the 2543bis draft. They seem
> to overlap to some large extent, so a decision needs to be reached as to
> whether they are truly different things or just one and, if one, which
> of the two to pick. Content-Disposition is an existing concept (RFC
> 2183), while Content-Function is not used outside of SIP. My preference
> would be to stick with Content-Disposition.
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Neil Deason wrote:
> 
> This draft states that a probing mechanism should be defined to aid
> backwards compatibility when defining new methods. It describes the
> mechanism to be used as the UAC adding some header to a standard SIP
> request. The UAS places some information in the response if it
> understands this header (and thus the extension-method).
> 
> Couldn't the OPTIONS method be used to provide such a probing mechanism
> within the standard SIP framework? The response to an OPTIONS request
> should include an Allow header entity-field that lists the supported set
> of headers, including any extension-methods.

Well, Allow lists allowed methods, not new extensions. New extensions
are listed in the Supported header.

Since this draft was published, there was some consensus on the list
that the response to INVITE should probably contain both Allow and
Supported headers (and possibly even Accept). The particular application
was TRANSFER; it would be nice to know once the call is set up whether
the remote side can be transferred. This would allow you to "grey out"
the transfer button on the UI, rather than having the user try and then
fail.

This would eliminate the need for the probing mechanism, and avoid the
need for the explicit OPTIONS.

Suggested wording:

"A 200 OK response to the initial INVITE for a call SHOULD contain the
Allow and Supported headers, and MAY contain the Accept header. These
will enable the UAC to determine the features and extensions supported
by the UAS for the duration of the call. Similarly, the initial INVITE
from the UAC SHOULD contain the Allow and Supported headers, and MAY
contain the Accept header."

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Date: Fri, 26 May 2000 10:23:13 +0200 (CEST)
From: Lars Berggren <lars@intertex.se>
To: Alberto Conte <Alberto.Conte@ms.alcatel.fr>
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Neil Deason <ndeason@ubiquity.net>, IETF SIP <sip@lists.bell-labs.com>
Subject: Re: [SIP] INVITE with incomplete SDP description
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On Thu, 25 May 2000, Alberto Conte wrote:

> Putting in hold the calle is a solution, but that increases the number of messages to
> be exchanged, requiring a re-INVITE transaction!
> 
> The power of SIP is to reduce the message exchanged in case of no error. And this is
> actually a case of no-error. It should be better allowing the caller to propose its
> media request delaying only the RTP port information in the ACK.
> 
> Alberto 
> 

I would rather think the power of SIP is its generality, flexibility and
simplicity. A re-INVITE is a very general and powerful mechanism to
accomplish many different tasks, including yours, so why not keep it that
way instead of specifying special cases which tend to make the protocol
more complex.

/Lars

Lars Berggren       <mailto:lars.berggren@intertex.se>
Intertex Data AB    tel: +46-8-6282828
Sundbyberg, Sweden  fax: +46-8-6286414




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Thanks for your reply Jonathan. Comments inline...

Jonathan Rosenberg wrote:

> Neil Deason wrote:
> >
> > This draft states that a probing mechanism should be defined to aid
> > backwards compatibility when defining new methods. It describes the
> > mechanism to be used as the UAC adding some header to a standard SIP
> > request. The UAS places some information in the response if it
> > understands this header (and thus the extension-method).
> >
> > Couldn't the OPTIONS method be used to provide such a probing mechanism
> > within the standard SIP framework? The response to an OPTIONS request
> > should include an Allow header entity-field that lists the supported set
> > of headers, including any extension-methods.
>
> Well, Allow lists allowed methods, not new extensions. New extensions
> are listed in the Supported header.

I am not sure I see your point - the probing mechanism defined in the draft is
described only in relation to new methods not extensions. But the response to
an OPTIONS should include a Supported header. This may all be academic though
given the proposed solution ...

> Since this draft was published, there was some consensus on the list
> that the response to INVITE should probably contain both Allow and
> Supported headers (and possibly even Accept). The particular application
> was TRANSFER; it would be nice to know once the call is set up whether
> the remote side can be transferred. This would allow you to "grey out"
> the transfer button on the UI, rather than having the user try and then
> fail.
>
> This would eliminate the need for the probing mechanism, and avoid the
> need for the explicit OPTIONS.

> Suggested wording:
>
> "A 200 OK response to the initial INVITE for a call SHOULD contain the
> Allow and Supported headers, and MAY contain the Accept header. These
> will enable the UAC to determine the features and extensions supported
> by the UAS for the duration of the call. Similarly, the initial INVITE
> from the UAC SHOULD contain the Allow and Supported headers, and MAY
> contain the Accept header."

What if an extension method was an "outside of the call" method, so there was
no preceeding INVITE. Is it sufficient just to send the new method and accept
the response you get might be 405 Method Not Allowed?

Cheers,
Neil.
--
Ubiquity Software Corporation, UK           http://www.ubiquity.net



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Hullo,

Jonathan Rosenberg wrote:
> To date, OPTIONS has been one of those "outside of the call" methods
> which has no impact on call state. Changing the address for routing
> subsequent requests was something I thought was handled through
> re-INVITE; having this also happen through OPTIONS seems to add another
> thing to check for, and also means that OPTIONS now has an effect on
> calls, when it didn't before.

Agreed.  If a Contact header in an OPTIONS request/response could
affect message routing, wouldn't this imply a potential need to
add a Record-Route header?

Actually, that reminds me: at the Bake-Off, there was some discussion
about the use of Route/Record-Route; I think the following points were
raised:
    a) any old SIP Entity can "force" a Proxy to send a message
       anywhere, just by inserting a Route header;
    b) does Record-Route only apply to the next Request (I think
       that this is now confirmed by the bis draft);
    c) is there any way to use the Record-Route mechanism to
       store state.

I guess a) can be solved by Authentication, or similar, but it
could also be solved by c).  Has there been any sort of progress/
thoughts/whatever on this at all?

Also, what is the motivation behind:
    "It [Record-Route] is also added by any proxy that translates
     between TCP and UDP or vice versa."
(Section 6.34.1 of May bis)

Cheers,


 - Jo.



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To: "Michael A. Ramalho" <mramalho@cisco.com>
From: Romeo Zwart <Romeo.Zwart@icoe.att.com>
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James/Mike/Mike, et al.

At 04:36 PM 22-05-00 -0400, Michael A. Ramalho wrote:
 > James/Mike,
 > I, for one, am grateful that you two have realized that
 > this is the crux of the debate between you. Can we go
 > on now?

The main disconnect in this discussion seems to be
the mixing of issues. On the one hand there's talk about
standardization of _services_ which is, as Mike mentioned
below, unlikely to be successful. In fact, I would argue that
even most "Old Word" operators realize that their strength
should be in service differentation, not in locking down
users into a handful of predefined features that everyone
offers.

On the other hand, there is the idea of having a mechanism
to exchange service _profiles_. This allows service portability
between service providers (whether Old or New World
operators), but brings with it a lot of issues around execution
control, relyability, etc. Defining a(nother) service definition
language, be that based on XML or anything else, would
have to take into account most if not all of the argumentation
that lead to CPL.

 > ... [stuff yanked] ...>
 > However, it is *very* clear to me that in the "new world"
 > of web-based or web-like services (and services definition)
 > that standardization of anything but the most rudimentary
 > subscriber information or core services will not make it in
 > "new world" deployments.

Fully agree with that.

 > Don't fence me in,
 >
 > Michael Ramalho
 > ... [more stuff yanked] ...
 >

Romeo 


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Sorry, but I still disagree. 

I don't think that specifying what does it mean a port number = 0 in the INVITE
message (which is at the moment not defined in the RFC 2543) is a way to make the
protocol more complex. 

I think it is just a way to ensure implementation compatibility by specifying how
this kind of message must be handled. That will never prevent to use the solution
proposed by Jonathan.

Alberto

Lars Berggren a écrit :
> 
> On Thu, 25 May 2000, Alberto Conte wrote:
> 
> > Putting in hold the calle is a solution, but that increases the number of messages to
> > be exchanged, requiring a re-INVITE transaction!
> >
> > The power of SIP is to reduce the message exchanged in case of no error. And this is
> > actually a case of no-error. It should be better allowing the caller to propose its
> > media request delaying only the RTP port information in the ACK.
> >
> > Alberto
> >
> 
> I would rather think the power of SIP is its generality, flexibility and
> simplicity. A re-INVITE is a very general and powerful mechanism to
> accomplish many different tasks, including yours, so why not keep it that
> way instead of specifying special cases which tend to make the protocol
> more complex.
> 
> /Lars
> 
> Lars Berggren       <mailto:lars.berggren@intertex.se>
> Intertex Data AB    tel: +46-8-6282828
> Sundbyberg, Sweden  fax: +46-8-6286414


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Alberto Conte wrote:
> 
> Sorry, but I still disagree.
> 
> I don't think that specifying what does it mean a port number = 0 in the INVITE
> message (which is at the moment not defined in the RFC 2543) is a way to make the
> protocol more complex.

Unfortunately, what you are proposing to do with port=0 is different
from what it means in the response. That makes it more complex. Its also
not backwards compatible, since this is not supported in rfc2543.
However, using the reINVITE mechanism I propose is backwards compatible. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Subject: Re: [SIP] INVITE with incomplete SDP description
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I've been following this thread with interest because we're wrestling with a similar problem. So far I
like Jonathan's suggestion the best (setting "c" destination address to 0.0.0.0) because that seems to
be semantically consistent with the intent, i.e., start a session with a call leg (very briefly) on
hold. The only niggle is using a re-INVITE, which is a perfectly good solution in my mind, rather than
the ACK from the original INVITE. I note that the third party call control draft suggests that sending
the SDP in the ACK after an INVITE with no SDP content is OK, so I'm not sure why the same principal
wouldn't apply here. Am I missing something?

Rick Workman
Nortel Networks
Ottawa

Jonathan Rosenberg wrote:

> Another way to do it is to send the initial INVITE with the desired
> media streams on hold (IP address set to 0.0.0.0). When you have
> specific IP address and port information, you can do a re-INVITE, and
> take the user off hold by passing an updated IP address and port in each
> m line. Note that the update would not occur in an ACK, but rather in a
> re-INVITE.
>
> -Jonathan R.
>
> Neil Deason wrote:
> >
> > Alberto Conte wrote:
> >
> > > Lars Berggren a écrit :
> > > >
> >
> > <snip>
> >
> > > > In this case you can send an INVITE without SDP, the callee lists streams
> > > > in the 200, and the ACK can contain the final SDP. This behaviour is
> > > > referenced in the spec. /Lars
> > > >
> > >
> > > Sorry Lars, I don't agree with you.
> > >
> > > Sending an INVITE without SDP means that the caller will be forced to choose the
> > > medias exchanged in the session on the base of the ones proposed by the callee!
> > >
> > > This is not exactly what I want. I would like the caller able to force the callee to
> > > accept its choice (or a part) of the medias, delaying only the RTP port and address
> > > information.
> >
> > But by specifying the medias in an INVITE you don't 'force' the callee to accept the caller's
> > choice. It is the opening offer in the two parties agreeing on media types. If you do not include
> > the session description in the INVITE, the other party should offer it's supported media streams
> > and codecs in its 1xx or 2xx response. Your UA then sends a completed session description in the
> > ACK.
> >
> > Cheers,
> > Neil.
> > --
> > Ubiquity Software Corporation, UK           http://www.ubiquity.net
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
>
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
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Subject: Re: [SIP] INVITE with incomplete SDP description
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Jonathan Rosenberg a écrit :
> 
> Alberto Conte wrote:
> >
> > Sorry, but I still disagree.
> >
> > I don't think that specifying what does it mean a port number = 0 in the INVITE
> > message (which is at the moment not defined in the RFC 2543) is a way to make the
> > protocol more complex.
> 
> Unfortunately, what you are proposing to do with port=0 is different
> from what it means in the response. That makes it more complex. Its also
> not backwards compatible, since this is not supported in rfc2543.
> However, using the reINVITE mechanism I propose is backwards compatible.
> 
> -Jonathan R.
> 

My point is: which is the behavior of a RFC2543 compliant SIP terminal when receiving
an INVITE message with the format that I proposed? 

I don't think this is defined. I think to something like "400 Bad Request" .

In such case I don't see any problem of backward compatibility. Terminals not
supporting the port=0 format should always reply with that error message. 

Alberto


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Subject: Re: [SIP] Contact in OPTIONS
To: jhornsby@ubiquity.net (Jo Hornsby)
Date: Fri, 26 May 2000 10:14:12 -0500 (CDT)
Cc: jdrosen@dynamicsoft.com (Jonathan Rosenberg),
        schulzrinne@cs.columbia.edu (Henning Schulzrinne),
        sip@lists.bell-labs.com
In-Reply-To: <001901bfc6f0$82b7d680$4e34c3c1@ubiquity.co.uk> from "Jo Hornsby" at May 26, 2000 09:58:07 AM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>Actually, that reminds me: at the Bake-Off, there was some discussion
>about the use of Route/Record-Route; I think the following points were
>raised:
>    a) any old SIP Entity can "force" a Proxy to send a message
>       anywhere, just by inserting a Route header;
>    b) does Record-Route only apply to the next Request (I think
>       that this is now confirmed by the bis draft);
>    c) is there any way to use the Record-Route mechanism to
>       store state.
>
>I guess a) can be solved by Authentication, or similar, but it
>could also be solved by c).  Has there been any sort of progress/
>thoughts/whatever on this at all?

Actually, I thought B was somewhat controversial; I'm surprised
that it was changed without a lot of discussion on the list.
Personally, I have no objection to the new behaviour; I just
severely dislike the fact that it is not backwards compatible
with the 2543 draft. Specifically, 2543-based call-stateful
proxies (including firewalls) will break in all kinds of ways
with 2543bis-based clients, if this change is finalized. Yuck.

Perhaps we can add something to the message so that 2543bis based 
clients can tell that the "new" behavior is being used?

>Also, what is the motivation behind:
>    "It [Record-Route] is also added by any proxy that translates
>     between TCP and UDP or vice versa."
>(Section 6.34.1 of May bis)

Presumably, this function (TCP/UDP translation) would be provided
only if the clients couldn't talk to each other directly (e.g.
one is TCP only, and the other is UDP only). In this case, the
proxy must remain in the path to continue to provide TCP/UDP
translation for the duration of the call leg.

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


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Subject: Re: [SIP] INVITE with incomplete SDP description
To: Alberto.Conte@ms.alcatel.fr
Date: Fri, 26 May 2000 10:30:44 -0500 (CDT)
Cc: jdrosen@dynamicsoft.com (Jonathan Rosenberg),
        ndeason@ubiquity.net (Neil Deason), lars@intertex.se (Lars Berggren),
        sip@lists.bell-labs.com (IETF SIP)
In-Reply-To: <392D4F2A.9571F52F@ms.alcatel.fr> from "Alberto Conte" at May 25, 2000 06:04:58 PM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>Putting in hold the calle is a solution, but that increases the number of messages to
>be exchanged, requiring a re-INVITE transaction!
>
>The power of SIP is to reduce the message exchanged in case of no error. And this is
>actually a case of no-error. It should be better allowing the caller to propose its
>media request delaying only the RTP port information in the ACK.

I'm going to have to go along with Lars and Jonathan here.
The power of SIP is in its generality. The tools defined with
the protocol as it stands today are flexible enough to allow
solutions to a wide variety of problems, including yours,
without requiring protocol changes (and complicated exception
cases in implementations) to handle special-case optimisations
like the one you propose.

For quite some time now, the SIP working group has been operating
under the premise that generality is far superior to optimisations.
It is an explicitly stated working group doctrine. I don't beleive
the situation you are describing is in any way special enough to
radically change our mode of operation.

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


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Subject: Re: [SIP] Error in "SIP Call Control Services"? - was [SIP]Multimedia Conferencing
To: hyongsop@research.telcordia.com (Hyong Sop Shim)
Date: Fri, 26 May 2000 10:18:49 -0500 (CDT)
Cc: jdrosen@dynamicsoft.com (Jonathan Rosenberg),
        sip@lists.research.bell-labs.com
In-Reply-To: <392D9F75.278396A7@research.telcordia.com> from "Hyong Sop Shim" at May 25, 2000 05:47:33 PM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>In Figure 4 in 8.3 Add Party during Add Party, D, acting on the untriggered INVITE
>message from B, sends an INVITE message to A, "INV A Also:A, B."  But, shouldn't
>this INVITE message from D be "INV A Also:B ReqBy:B" instead?

This draft has expired; I don't beleive a new version will be issued.

The way forward with call-control which will probably be adopted
is outlined in draft-campbell-sip-cc-framework-00.txt (which
is being updated to bring it in line with 
draft-rosenberg-sip-guidelines-00.txt). An example of the modular
approach espoused in this draft is presented in 
draft-sparks-sip-cc-transfer-00.txt (which will probably be changing 
to be more general in its next release).

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


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To: <sip@lists.bell-labs.com>
Subject: RE: [SIP] Contact in OPTIONS
Date: Fri, 26 May 2000 16:47:31 +0100
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> >Also, what is the motivation behind:
> >    "It [Record-Route] is also added by any proxy that translates
> >     between TCP and UDP or vice versa."
> >(Section 6.34.1 of May bis)
> 
> Presumably, this function (TCP/UDP translation) would be provided
> only if the clients couldn't talk to each other directly (e.g.
> one is TCP only, and the other is UDP only). In this case, the
> proxy must remain in the path to continue to provide TCP/UDP
> translation for the duration of the call leg.

Yes.  A colleague (hi James!) suggested much the same.

It does potentially seem to be a little bit of a shame, however;
I can imagine a chain of proxies presenting a path that alternates
between UDP/TCP with User Agents talking UDP (or TCP) at either end.

Of course, I'm at a loss for anything better.

Ah well,


 - Jo.



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From: "Vijay K. Gurbani" <vkg@lucent.com>
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Subject: Re: [SIP] Contact in OPTIONS
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"Adam B. Roach" wrote:
> 
> >Actually, that reminds me: at the Bake-Off, there was some discussion
> >about the use of Route/Record-Route; I think the following points 
> >were raised:
> >    a) any old SIP Entity can "force" a Proxy to send a message
> >       anywhere, just by inserting a Route header;
> >    b) does Record-Route only apply to the next Request (I think
> >       that this is now confirmed by the bis draft);
> >    c) is there any way to use the Record-Route mechanism to
> >       store state.
[...]
> Actually, I thought B was somewhat controversial; I'm surprised
> that it was changed without a lot of discussion on the list.
> Personally, I have no objection to the new behaviour; I just
> severely dislike the fact that it is not backwards compatible
> with the 2543 draft. Specifically, 2543-based call-stateful
> proxies (including firewalls) will break in all kinds of ways
> with 2543bis-based clients, if this change is finalized. Yuck.

I agree with Adam here.  RFC2543bis does state that this "...field is
added to a request by any proxy that insists on being in the path of
the next request for the same call leg".  I liked the RFC2543 behavior
better when the field was added to a request by any proxy that insists
on being in the path of _subsequent_ requests.  This will break existing
proxy behavior.

I am sure there are reasons why the per-request model was chosen in 
RFC2543bis?  Could someone refresh my memory with the most pressing 
ones?

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
IN Architecture and Internet Software Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184


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Subject: Re: [SIP] Contact in OPTIONS
To: vkg@lucent.com
Date: Fri, 26 May 2000 12:25:54 -0500 (CDT)
Cc: sip@lists.bell-labs.com
In-Reply-To: <392E9E6F.C2AD1B9C@lucent.com> from "Vijay K. Gurbani" at May 26, 2000 10:55:27 AM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>RFC2543bis does state that this "...field is
>added to a request by any proxy that insists on being in the path of
>the next request for the same call leg".  I liked the RFC2543 behavior
>better when the field was added to a request by any proxy that insists
>on being in the path of _subsequent_ requests.

This just jogged my memory about something else that might need
to be addressed in the case that "Record-Route" applies to only 
the next transaction, somewhat akin to re-INVITE glare. 

We need to closely examine the implications of requests crossing
on the wire; under these circumstances, the "next" transaction
is something of an ambiguous concept. I beleive this could lead
to some pretty spectacular headaches. In particular, think
of two clients with two proxies between them. If requests
cross between the two proxies... my head hurts thinking
about trying to reconcile the record-route state of all
four nodes.

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


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Adam B. Roach wrote:
> 
> >Actually, that reminds me: at the Bake-Off, there was some discussion
> >about the use of Route/Record-Route; I think the following points were
> >raised:
> >    a) any old SIP Entity can "force" a Proxy to send a message
> >       anywhere, just by inserting a Route header;
> >    b) does Record-Route only apply to the next Request (I think
> >       that this is now confirmed by the bis draft);
> >    c) is there any way to use the Record-Route mechanism to
> >       store state.
> >
> >I guess a) can be solved by Authentication, or similar, but it
> >could also be solved by c).  Has there been any sort of progress/
> >thoughts/whatever on this at all?
> 
> Actually, I thought B was somewhat controversial; I'm surprised
> that it was changed without a lot of discussion on the list.
> Personally, I have no objection to the new behaviour; I just
> severely dislike the fact that it is not backwards compatible
> with the 2543 draft. Specifically, 2543-based call-stateful
> proxies (including firewalls) will break in all kinds of ways
> with 2543bis-based clients, if this change is finalized. Yuck.

Consider the inclusion in 2543bis as an incentive to reach closure. It
might help to provide real examples where Record-Route is essential and
whether they'd be better off with next-request or rest-of-call
semantics. Firewalls clearly fall into the rest-of-call class.


> 
> Perhaps we can add something to the message so that 2543bis based
> clients can tell that the "new" behavior is being used?
> 
> >Also, what is the motivation behind:
> >    "It [Record-Route] is also added by any proxy that translates
> >     between TCP and UDP or vice versa."
> >(Section 6.34.1 of May bis)
> 
> Presumably, this function (TCP/UDP translation) would be provided
> only if the clients couldn't talk to each other directly (e.g.
> one is TCP only, and the other is UDP only). In this case, the
> proxy must remain in the path to continue to provide TCP/UDP
> translation for the duration of the call leg.

Since then, some additional discussion seems to indicate that things get
a lot easier if we require UAs to speak UDP as the lingua franca.
Otherwise, we can't satisfy the goal of simply plugging in a bunch of
SIP phones and setting up calls, without a proxy.


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From sip-admin@lists.bell-labs.com  Fri May 26 14:42:13 2000
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Henning Schulzrinne wrote:
[...]
> Consider the inclusion in 2543bis as an incentive to reach closure. It
> might help to provide real examples where Record-Route is essential 
> and whether they'd be better off with next-request or rest-of-call
> semantics. Firewalls clearly fall into the rest-of-call class.
[...]

As do call-model mapping proxies; i.e. SIP proxies which provide 
services based on a mapping of foreign call models.  Some of the work
we have done in this domain takes advantage of the "rest-of-call"
semantics by having the proxy inject a Record-Route in the first INVITE 
message and being guaranteed that subsequent requests (ACK, CANCEL or 
BYE) and responses (especially 200 OK) will come through it.

Regards,

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
IN Architecture and Internet Software Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184


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From sip-admin@lists.bell-labs.com  Fri May 26 20:23:42 2000
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Subject: [SIP] SIP Application Server
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Hi, All

I am writing a design document for a SIP network

Is Application Server "a kind of" Proxy Server or Redirect Server or
Neither?
(Application Server will provide some services)




Thank you so much

Tricia




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From sip-admin@lists.bell-labs.com  Sat May 27 00:51:31 2000
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My understanding of an 'Application Server' is that it is a logical entity,
that provides services to network entities. What services depends on what
Applications are executing in this App. Server and what entity is being
interfaced with.

If you now talk about the SIP Domain, whether the Application Server is
colocated 'physically' with a Proxy server or some other SIP entity is an
implementation issue.

But my guess would be that a SIP Proxy is likely to interface with your
logical Application Server, since it would usually be providing the most
services. That does not however imply that the App. Server is necessarily
sitting on top of a proxy /redirect.


In short, as far as I know,  Application Server is a logical entity that
*can* be physically colocated with your SIP Proxy, if you so wish - or it
can be a separate Physical server with some means of communication with all
entities requesting services (the entities can be Proxies, terminal , or
any other such device)

Regds
Arjun

--
Arjun Roychowdhury @ Hughes Software Systems









Tricia Wang <TWang@NeTrue.com> on 05/27/2000 05:55:26 AM

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

Subject:  [SIP] SIP Application Server




Hi, All

I am writing a design document for a SIP network

Is Application Server "a kind of" Proxy Server or Redirect Server or
Neither?
(Application Server will provide some services)




Thank you so much

Tricia




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From sip-admin@lists.bell-labs.com  Sat May 27 23:13:53 2000
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Subject: Re: [SIP] Contact in OPTIONS
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Another application I am aware of for record-routing, generation of call
detail records for proxies fronting a gateway, also requires it to see
all requests.

Seems like consensus so far to keep the current semantics.... anyone
object?

Thanks,
Jonathan R.

"Vijay K. Gurbani" wrote:
> 
> Henning Schulzrinne wrote:
> [...]
> > Consider the inclusion in 2543bis as an incentive to reach closure. It
> > might help to provide real examples where Record-Route is essential
> > and whether they'd be better off with next-request or rest-of-call
> > semantics. Firewalls clearly fall into the rest-of-call class.
> [...]
> 
> As do call-model mapping proxies; i.e. SIP proxies which provide
> services based on a mapping of foreign call models.  Some of the work
> we have done in this domain takes advantage of the "rest-of-call"
> semantics by having the proxy inject a Record-Route in the first INVITE
> message and being guaranteed that subsequent requests (ACK, CANCEL or
> BYE) and responses (especially 200 OK) will come through it.
> 
> Regards,
> 
> - vijay
> --
> Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
> IN Architecture and Internet Software Group
> Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
> Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Henning Schulzrinne wrote:
> 
> > Presumably, this function (TCP/UDP translation) would be provided
> > only if the clients couldn't talk to each other directly (e.g.
> > one is TCP only, and the other is UDP only). In this case, the
> > proxy must remain in the path to continue to provide TCP/UDP
> > translation for the duration of the call leg.
> 
> Since then, some additional discussion seems to indicate that things get
> a lot easier if we require UAs to speak UDP as the lingua franca.
> Otherwise, we can't satisfy the goal of simply plugging in a bunch of
> SIP phones and setting up calls, without a proxy.


To be specific, the proposal is that UDP support in UA transition from
SHOULD to MUST, so that:

1. UAs SHOULD support TCP, MUST support UDP
2. proxies MUST support TCP and UDP

The transport parameter in the Contact and Record-Route headers then
becomes the preferential choice; if not supported, you can always fall
back to UDP.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: [SIP] draft-rosenberg-sip-guidelines-00.txt
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Neil Deason wrote:
> 
> > > Couldn't the OPTIONS method be used to provide such a probing mechanism
> > > within the standard SIP framework? The response to an OPTIONS request
> > > should include an Allow header entity-field that lists the supported set
> > > of headers, including any extension-methods.
> >
> > Well, Allow lists allowed methods, not new extensions. New extensions
> > are listed in the Supported header.
> 
> I am not sure I see your point - the probing mechanism defined in the draft is
> described only in relation to new methods not extensions. 

Right, thats my point. Your comment was:

> The response to an OPTIONS request
> should include an Allow header entity-field that lists the supported set
> of headers, including any extension-methods.

which was that the Allow header should list supported headers. It does
not. It only lists allowed methods.


> What if an extension method was an "outside of the call" method, so there was
> no preceeding INVITE. Is it sufficient just to send the new method and accept
> the response you get might be 405 Method Not Allowed?

Of course. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Sun May 28 04:05:03 2000
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Folks,

I've just submitted an updated version of the I-D on "Reliability of
Provisional Responses in SIP". Until it appears in the archives, you can
find a copy at:

http://www.cs.columbia.edu/~jdrosen/papers/draft-ietf-sip-100rel-01.txt

The changes since the last version are:

1. CANCEL disallowed for PRACK
2. Require allowed in requests for this extension
3. option tag changed from org.ietf.sip.100rel to 100rel
4. Proxy-Require disallowed
5. discussion of sending final responses to original request before all
reliable provisional responses have been received (its allowed)
6. clarifications

Once this draft appears in the archives, a two week working group last
call will begin. At the end of that last call, if no comments are heard,
the document will be sent to IESG for consideration as proposed
standard.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Mon May 29 05:44:54 2000
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Date: Mon, 29 May 2000 11:46:21 +0200 (CEST)
From: Lars Berggren <lars@intertex.se>
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Cc: "Adam B. Roach" <Adam.Roach@Ericsson.com>,
        Jo Hornsby <jhornsby@ubiquity.net>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@lists.bell-labs.com
Subject: Re: [SIP] Contact in OPTIONS
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On Fri, 26 May 2000, Henning Schulzrinne wrote:
> Adam B. Roach wrote:
> > 
> > Actually, I thought B was somewhat controversial; I'm surprised
> > that it was changed without a lot of discussion on the list.
> > Personally, I have no objection to the new behaviour; I just
> > severely dislike the fact that it is not backwards compatible
> > with the 2543 draft. Specifically, 2543-based call-stateful
> > proxies (including firewalls) will break in all kinds of ways
> > with 2543bis-based clients, if this change is finalized. Yuck.
> 
> Consider the inclusion in 2543bis as an incentive to reach closure. It
> might help to provide real examples where Record-Route is essential and
> whether they'd be better off with next-request or rest-of-call
> semantics. Firewalls clearly fall into the rest-of-call class.
> 

I recall an earlier discussion on the list where arguements were made that
the next-request behaviour would help a UA that crashes, reboots and
receives a re-INVITE for the call. The INVITE is designed so that it 
carries the information necessary to join the session regardless of
whether you know about previous requests or not. The next-request
mechanism would adhere to that design decision.

Also, in favour of the next-request behaviour, if the route can be changed
as the call progresses, I guess it can be useful. A proxy might want to
step out of the signalling path for example, but I don't have any real
world example on this.

However I also dislike the fact that the next-request mechanism is not
backwards compatible.

/Lars

Lars Berggren       <mailto:lars.berggren@intertex.se>
Intertex Data AB    tel: +46-8-6282828
Sundbyberg, Sweden  fax: +46-8-6286414



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From sip-admin@lists.bell-labs.com  Mon May 29 09:35:17 2000
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Lars Berggren wrote:
> 
> On Fri, 26 May 2000, Henning Schulzrinne wrote:
> > Adam B. Roach wrote:
> > >
> > > Actually, I thought B was somewhat controversial; I'm surprised
> > > that it was changed without a lot of discussion on the list.
> > > Personally, I have no objection to the new behaviour; I just
> > > severely dislike the fact that it is not backwards compatible
> > > with the 2543 draft. Specifically, 2543-based call-stateful
> > > proxies (including firewalls) will break in all kinds of ways
> > > with 2543bis-based clients, if this change is finalized. Yuck.
> >
> > Consider the inclusion in 2543bis as an incentive to reach closure. It
> > might help to provide real examples where Record-Route is essential and
> > whether they'd be better off with next-request or rest-of-call
> > semantics. Firewalls clearly fall into the rest-of-call class.
> >
> 
> I recall an earlier discussion on the list where arguements were made that
> the next-request behaviour would help a UA that crashes, reboots and
> receives a re-INVITE for the call. The INVITE is designed so that it
> carries the information necessary to join the session regardless of
> whether you know about previous requests or not. The next-request
> mechanism would adhere to that design decision.
> 
> Also, in favour of the next-request behaviour, if the route can be changed
> as the call progresses, I guess it can be useful. A proxy might want to
> step out of the signalling path for example, but I don't have any real
> world example on this.
> 
> However I also dislike the fact that the next-request mechanism is not
> backwards compatible.

One possibility to increase robustness in the face of restarts might be
to have proxies insert Record-Route in every request. That way, the
rebooted component will still get the right information. This would seem
to be backward compatible. In this model, once a proxy adds itself to
the route, it cannot delete itself, i.e., *not* including does not imply
dropping out. Also, changing RR entries (for example, changing the maddr
part) could not be guaranteed to be respected by the other side.

Would this work?

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


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From sip-admin@lists.bell-labs.com  Mon May 29 13:29:43 2000
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From: Samandi Sami <Sami.Samandi@SRIT.siemens.fr>
To: "'archow@hss.hns.com'" <archow@hss.hns.com>,
        Tricia Wang <TWang@NeTrue.com>
Cc: sip@lists.bell-labs.com
Subject: RE: [SIP] SIP Application Server
Date: Mon, 29 May 2000 19:31:14 +0200
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Hello,

Is there any document (or work) out there describing the distribution of
logic between a feature server (or application server) and the underlying
network server asking for the services. It is true this is an implementation
issue but it is also true that this split of the roles between a SIP
application server and a SIP proxy server is similar to the IN architecture
where the SSP handles the call control and the SCP the service control.  Is
it then logical to think about an implementation like the following :

-	The network server is a call stateful proxy (holding the state of a
call between the INVITE and the BYE). This SIP proxy forward requests to the
application server. This proxy server goes stateful and proxy requests only
when needed (for instance when the end user has subscribed to predefined
services).
-	The application server will behave per call basis whether as a
redirect server (to provide routing services) or as a UAS when media
services are needed (IVR, ...). I am not sure whether this equipment should
also proxy the request ?.
-	The goal here is not to involve the application server for the whole
duration of the call but only when needed and this means telling the
application server what's state of the call that triggered the call of the
service. For instance, A call B, but  B is busy,  the Proxy server of B
receives a 486 and then sends an INVITE to the feature server asking what to
do. How to tell the application server that the user is busy (the equivalent
of the DP Terminal busy of the IN) ? Should this be done in a proprietary
way ?

Could someone comment on this please even if it seems straightforward and
maybe a quite high level views.

Thank you

Sami

	-----Original Message-----
	From:	archow@hss.hns.com [SMTP:archow@hss.hns.com]
	Sent:	samedi 27 mai 2000 06:59
	To:	Tricia Wang
	Cc:	sip@lists.bell-labs.com
	Subject:	Re: [SIP] SIP Application Server




	My understanding of an 'Application Server' is that it is a logical
entity,
	that provides services to network entities. What services depends on
what
	Applications are executing in this App. Server and what entity is
being
	interfaced with.

	If you now talk about the SIP Domain, whether the Application Server
is
	colocated 'physically' with a Proxy server or some other SIP entity
is an
	implementation issue.

	But my guess would be that a SIP Proxy is likely to interface with
your
	logical Application Server, since it would usually be providing the
most
	services. That does not however imply that the App. Server is
necessarily
	sitting on top of a proxy /redirect.


	In short, as far as I know,  Application Server is a logical entity
that
	*can* be physically colocated with your SIP Proxy, if you so wish -
or it
	can be a separate Physical server with some means of communication
with all
	entities requesting services (the entities can be Proxies, terminal
, or
	any other such device)

	Regds
	Arjun

	--
	Arjun Roychowdhury @ Hughes Software Systems









	Tricia Wang <TWang@NeTrue.com> on 05/27/2000 05:55:26 AM

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

	Subject:  [SIP] SIP Application Server




	Hi, All

	I am writing a design document for a SIP network

	Is Application Server "a kind of" Proxy Server or Redirect Server or
	Neither?
	(Application Server will provide some services)




	Thank you so much

	Tricia




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        Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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Subject: RE: [SIP] Contact in OPTIONS
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One example of when a proxy might want to step out of the signalling path is
that of overlap outpulsing in a gateway situation.  The proxy might stay in
the path until it had determined that all digits had been collected.  At
that point, having translated the digits to an ultimate destination, it
could retire from the call.  

> -----Original Message-----
> From:	Lars Berggren [SMTP:lars@intertex.se]
> Sent:	Monday, May 29, 2000 5:46 AM
> To:	Henning Schulzrinne
> Cc:	Adam B. Roach; Jo Hornsby; Jonathan Rosenberg;
> sip@lists.bell-labs.com
> Subject:	Re: [SIP] Contact in OPTIONS
> 
> On Fri, 26 May 2000, Henning Schulzrinne wrote:
> > Adam B. Roach wrote:
> > > 
> > > Actually, I thought B was somewhat controversial; I'm surprised
> > > that it was changed without a lot of discussion on the list.
> > > Personally, I have no objection to the new behaviour; I just
> > > severely dislike the fact that it is not backwards compatible
> > > with the 2543 draft. Specifically, 2543-based call-stateful
> > > proxies (including firewalls) will break in all kinds of ways
> > > with 2543bis-based clients, if this change is finalized. Yuck.
> > 
> > Consider the inclusion in 2543bis as an incentive to reach closure. It
> > might help to provide real examples where Record-Route is essential and
> > whether they'd be better off with next-request or rest-of-call
> > semantics. Firewalls clearly fall into the rest-of-call class.
> > 
> 
> I recall an earlier discussion on the list where arguements were made that
> the next-request behaviour would help a UA that crashes, reboots and
> receives a re-INVITE for the call. The INVITE is designed so that it 
> carries the information necessary to join the session regardless of
> whether you know about previous requests or not. The next-request
> mechanism would adhere to that design decision.
> 
> Also, in favour of the next-request behaviour, if the route can be changed
> as the call progresses, I guess it can be useful. A proxy might want to
> step out of the signalling path for example, but I don't have any real
> world example on this.
> 
> However I also dislike the fact that the next-request mechanism is not
> backwards compatible.
> 
> /Lars
> 
> Lars Berggren       <mailto:lars.berggren@intertex.se>
> Intertex Data AB    tel: +46-8-6282828
> Sundbyberg, Sweden  fax: +46-8-6286414
> 
> 

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<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">One example of when =
a proxy might want to step out of the signalling path is that of =
overlap outpulsing in a gateway situation.&nbsp; The proxy might stay =
in the path until it had determined that all digits had been =
collected.&nbsp; At that point, having translated the digits to an =
ultimate destination, it could retire from the call.&nbsp; </FONT></P>
<UL>
<P><FONT SIZE=3D1 FACE=3D"Arial">-----Original Message-----</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">From:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Lars Berggren [SMTP:lars@intertex.se]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Monday, May 29, 2000 5:46 AM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">Henning Schulzrinne</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Cc:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">Adam B. Roach; Jo Hornsby; Jonathan Rosenberg; =
sip@lists.bell-labs.com</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT>=
</B> <FONT SIZE=3D1 FACE=3D"Arial">Re: [SIP] Contact in OPTIONS</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">On Fri, 26 May 2000, Henning =
Schulzrinne wrote:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Adam B. Roach wrote:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; Actually, I thought B was =
somewhat controversial; I'm surprised</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; that it was changed without =
a lot of discussion on the list.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; Personally, I have no =
objection to the new behaviour; I just</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; severely dislike the fact =
that it is not backwards compatible</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; with the 2543 draft. =
Specifically, 2543-based call-stateful</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; proxies (including =
firewalls) will break in all kinds of ways</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; with 2543bis-based clients, =
if this change is finalized. Yuck.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Consider the inclusion in =
2543bis as an incentive to reach closure. It</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; might help to provide real =
examples where Record-Route is essential and</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; whether they'd be better off =
with next-request or rest-of-call</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; semantics. Firewalls clearly =
fall into the rest-of-call class.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I recall an earlier discussion on the =
list where arguements were made that</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">the next-request behaviour would help =
a UA that crashes, reboots and</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">receives a re-INVITE for the call. =
The INVITE is designed so that it </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">carries the information necessary to =
join the session regardless of</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">whether you know about previous =
requests or not. The next-request</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">mechanism would adhere to that design =
decision.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Also, in favour of the next-request =
behaviour, if the route can be changed</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">as the call progresses, I guess it =
can be useful. A proxy might want to</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">step out of the signalling path for =
example, but I don't have any real</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">world example on this.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">However I also dislike the fact that =
the next-request mechanism is not</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">backwards compatible.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">/Lars</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Lars =
Berggren&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;<U></U></FONT><U><FONT =
COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"mailto:lars.berggren@intertex.se">mailto:lars.berggren@intertex.=
se</A></FONT></U><FONT SIZE=3D2 FACE=3D"Arial">&gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Intertex Data AB&nbsp;&nbsp;&nbsp; =
tel: +46-8-6282828</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Sundbyberg, Sweden&nbsp; fax: =
+46-8-6286414</FONT>
</P>
<BR>
</UL>
</BODY>
</HTML>
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From sip-admin@lists.bell-labs.com  Mon May 29 16:22:46 2000
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Date: Mon, 29 May 2000 15:15:04 -0500
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Subject: [SIP] Basic questions
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Hello, all. Just starting with SIP, but I already have a few basic questions
regarding the RFC. Since this is my first query, let me know if these
questions should be posted elsewhere or I have otherwise violated some other
rule of etiquette.

- "...SIP...is...for...sessions with one or more participants." - How can
there be a session with only one participant? A session is defined elsewhere
as comprising one or more RTP sessions, which implies a sender and a
receiver, or two participants. Should the sentence have been written as
"...with two or more participants?"

- "...the H.245 [8] gateway and user address and then use H.225.0 [9] to
establish the call." - What is meant by an "H.245 gateway address" and an
"H.245 user address?" There is no such thing. I assume that the author is
referring to the H.323 call-signaling addresses of a gateway and endpoint.
There is an H.245 control-protocol address, but if a separate H.245 channel
is even used, the address is conveyed during call establishment and not
before.

- The word, "program,: is used throughout the RFC. Is this any functional
entity regardless of implementation and not exclusively a "software
program?"

- "Outbound proxy" - Is there also such a thing as an inbound proxy?

- Regarding UA, UAC, and UAS, can this change for an entity on a per-call
basis? Can this change for an entity during a call?

- "Servers are either proxy, redirect or user agent servers or
registrars." - I have also seen references to "location servers" and "SIP
servers." Is a location server the same thing as a redirect server? Is "SIP
server" just a general reference to servers in "SIPdom?"

- Is "server" only a role played by an entity in a single request-response
dialogue, or are some entities always servers or always clients or always
both?

Paul Long
Smith micro Software, Inc.


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Subject: Re: [SIP] Contact in OPTIONS
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Henning Schulzrinne wrote:
> 
> > I recall an earlier discussion on the list where arguements were made that
> > the next-request behaviour would help a UA that crashes, reboots and
> > receives a re-INVITE for the call. The INVITE is designed so that it
> > carries the information necessary to join the session regardless of
> > whether you know about previous requests or not. The next-request
> > mechanism would adhere to that design decision.
> >
> > Also, in favour of the next-request behaviour, if the route can be changed
> > as the call progresses, I guess it can be useful. A proxy might want to
> > step out of the signalling path for example, but I don't have any real
> > world example on this.
> >
> > However I also dislike the fact that the next-request mechanism is not
> > backwards compatible.
> 
> One possibility to increase robustness in the face of restarts might be
> to have proxies insert Record-Route in every request. That way, the
> rebooted component will still get the right information. This would seem
> to be backward compatible. In this model, once a proxy adds itself to
> the route, it cannot delete itself, i.e., *not* including does not imply
> dropping out. Also, changing RR entries (for example, changing the maddr
> part) could not be guaranteed to be respected by the other side.
> 
> Would this work?

This definitely works. Basically, what it means is that so long as a UA
has state for the call, it keeps the original route and does not try to
update. If, however, it crashes and restarts, it uses whatever it gets
in the request. This would allow it to do a partial reconstruction of
the route - "better than nothing" kind of thing, with increasing
performance as this feature is spread.

In addition to this, we can define an explicit "remove me" record-route
entry. Something like:

Remove-Route: <sip:user@host;maddr=1.2.3.4>;remove=true

Proxies would add this when they want out. A UA not understanding this
will ignore out continue to use the old route, but a new UA would know
that this means to remove it from the Route in the next request. I think
this is also backwards compatible. 

Both of these mechanisms combined (remove-route and adding record-route
for mid-call messages) has the desired result of being backwards
compatible, providing for restarts after crashes, and allowing proxies
to remove themselves during a call.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Mon May 29 22:07:03 2000
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> Tom-PT Taylor wrote:
> 
> One example of when a proxy might want to step out of the signalling
> path is that of overlap outpulsing in a gateway situation.  The proxy
> might stay in the path until it had determined that all digits had
> been collected.  At that point, having translated the digits to an
> ultimate destination, it could retire from the call.

Careful here; the way overlap is currently designed, the proxy would not
have any choice about dropping out or staying in until the call
completes. Record-Route takes effect only after the 200 OK. Thus, in
your example, the proxy can elect to receive all of the overlap digits
and then "drop out" by simply never inserting record-route in the first
place. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Samandi Sami wrote:
> 
> Hello,
> 
> Is there any document (or work) out there describing the distribution of
> logic between a feature server (or application server) and the underlying
> network server asking for the services. It is true this is an implementation
> issue but it is also true that this split of the roles between a SIP
> application server and a SIP proxy server is similar to the IN architecture
> where the SSP handles the call control and the SCP the service control. 

The difference between and applications server and a proxy is only in
the complexity of the programs which guide their operation.

Both speak SIP; both can proxy or redirect; there is no way to
distinguish a proxy from an applications server from the outside. This
is quite different from the SSP/SCP model, which presumes totally
different interfaces and roles. I think the SIP model is a very powerful
model. It means that decisions about service invocation can always be
represented as routing decisions. If I want some service from provider
X, all that needs to happen is to have my call routed to the server at
domain X. 

One of the reasons the separation between call control and service
control was made in the IN, was the desire for the SCP not to receive
all signaling messages for the call. I will note that in SIP, this
function is already provided in the call signaling itself, through the
Record-Route mechanism. This means an application server can drop out of
the call after routing the initial request, as can a proxy. 

 Is
> it then logical to think about an implementation like the following :
> 
> -       The network server is a call stateful proxy (holding the state of a
> call between the INVITE and the BYE). This SIP proxy forward requests to the
> application server. This proxy server goes stateful and proxy requests only
> when needed (for instance when the end user has subscribed to predefined
> services).

Why? As I argue above, this distinction of an SSP/SCP role makes less
sense in SIP.

> -       The application server will behave per call basis whether as a
> redirect server (to provide routing services) or as a UAS when media
> services are needed (IVR, ...). I am not sure whether this equipment should
> also proxy the request ?.

If that is part of the service, sure.

> -       The goal here is not to involve the application server for the whole
> duration of the call but only when needed and this means telling the
> application server what's state of the call that triggered the call of the
> service. For instance, A call B, but  B is busy,  the Proxy server of B
> receives a 486 and then sends an INVITE to the feature server asking what to
> do. How to tell the application server that the user is busy (the equivalent
> of the DP Terminal busy of the IN) ? Should this be done in a proprietary
> way ?

I would not do it this way. I would send the initial INVITE to the
feature server, and when the 486 response comes, have it go from there. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Plong@smithmicro.com wrote:
> 
> Hello, all. Just starting with SIP, but I already have a few basic questions
> regarding the RFC. Since this is my first query, let me know if these
> questions should be posted elsewhere or I have otherwise violated some other
> rule of etiquette.

Many common questions are answered in the FAQ:
http://www.cs.columbia.edu/~hgs/sip/faq.html

> 
> - "...SIP...is...for...sessions with one or more participants." - How can
> there be a session with only one participant? A session is defined elsewhere
> as comprising one or more RTP sessions, which implies a sender and a
> receiver, or two participants. Should the sentence have been written as
> "...with two or more participants?"

Mulicast sessions can have a single participant. Not that interesting,
but possible.

> 
> - "...the H.245 [8] gateway and user address and then use H.225.0 [9] to
> establish the call." - What is meant by an "H.245 gateway address" and an
> "H.245 user address?" There is no such thing. I assume that the author is
> referring to the H.323 call-signaling addresses of a gateway and endpoint.
> There is an H.245 control-protocol address, but if a separate H.245 channel
> is even used, the address is conveyed during call establishment and not
> before.

You are of course correct; this should be H.323 call signaling address.

> 
> - The word, "program,: is used throughout the RFC. Is this any functional
> entity regardless of implementation and not exclusively a "software
> program?"

Of course. 

> 
> - "Outbound proxy" - Is there also such a thing as an inbound proxy?

Not formally. That would be the proxy which receives a request where the
request URI is its own domain. This is the general proxy described in
the specification. Note that there can be many such proxies of each type
on a call request path.


> 
> - Regarding UA, UAC, and UAS, can this change for an entity on a per-call
> basis? Can this change for an entity during a call?

The role of UAC and UAS is on a request by request basis, NOT a call by
call basis. This is probably confusing from the spec, as much of it was
written considering the initial INVITE request, for which caller and UAC
are the same.

> 
> - "Servers are either proxy, redirect or user agent servers or
> registrars." - I have also seen references to "location servers" and "SIP
> servers." Is a location server the same thing as a redirect server? Is "SIP
> server" just a general reference to servers in "SIPdom?"

Location server is a non-SIP server, like an LDAP server or SQL
database. "SIP server" is any entity that receives requests (UAS, proxy,
redirect, registrar).

> 
> - Is "server" only a role played by an entity in a single request-response
> dialogue, or are some entities always servers or always clients or always
> both?

Same as for UAS/UAC - its on a per request basis. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Original-From: vkg@ans.ih.lucent.com (Vijay K Gurbani)
To: jdrosen@dynamicsoft.com, schulzrinne@cs.columbia.edu
Subject: Re: [SIP] Contact in OPTIONS
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Henning Schulzrinne wrote:
> to have proxies insert Record-Route in every request. That way, the
> rebooted component will still get the right information. This would seem
> to be backward compatible. In this model, once a proxy adds itself to
> the route, it cannot delete itself, i.e., *not* including does not imply
> dropping out. Also, changing RR entries (for example, changing the maddr
> part) could not be guaranteed to be respected by the other side.
> 
> Would this work?

Jonathan Rosenberg wrote:
> This definitely works. Basically, what it means is that so long as a UA
> has state for the call, it keeps the original route and does not try to
> update. If, however, it crashes and restarts, it uses whatever it gets
> in the request. This would allow it to do a partial reconstruction of
> the route - "better than nothing" kind of thing, with increasing
> performance as this feature is spread.
> 
> In addition to this, we can define an explicit "remove me" record-route
> entry. Something like:
> 
> Remove-Route: <sip:user@host;maddr=1.2.3.4>;remove=true
> 
> Proxies would add this when they want out. A UA not understanding this
> will ignore out continue to use the old route, but a new UA would know
> that this means to remove it from the Route in the next request. I think
> this is also backwards compatible. 
> 
> Both of these mechanisms combined (remove-route and adding record-route
> for mid-call messages) has the desired result of being backwards
> compatible, providing for restarts after crashes, and allowing proxies
> to remove themselves during a call.

Jonathan:

Are you proposing a new header called "Remove-Route" or a new parameter
("remove=true") to the Record-Route header?  If Remove-Route will be
present in the general headers, why the "remove=true" parameter?

But yes, this should work, although deployed proxies may still think
that if they have included "Record-Route" in a previous request, that's
all they have to do.

Thanks,

- vijay
--
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
IN Architecture and Internet Software Group 
Lucent Technologies/Bell Labs Innovations 263 Shuman Blvd., Rm 1A-418
Naperville, Illinois 60566     Voice: +1 630 224 0216   Fax: +1 630 713 0184


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From sip-admin@lists.bell-labs.com  Tue May 30 00:51:17 2000
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Vijay K Gurbani wrote:
> 
> > In addition to this, we can define an explicit "remove me" record-route
> > entry. Something like:
> >
> > Remove-Route: <sip:user@host;maddr=1.2.3.4>;remove=true
> >
> > Proxies would add this when they want out. A UA not understanding this
> > will ignore out continue to use the old route, but a new UA would know
> > that this means to remove it from the Route in the next request. I think
> > this is also backwards compatible.
> >
> > Both of these mechanisms combined (remove-route and adding record-route
> > for mid-call messages) has the desired result of being backwards
> > compatible, providing for restarts after crashes, and allowing proxies
> > to remove themselves during a call.
> 
> Jonathan:
> 
> Are you proposing a new header called "Remove-Route" or a new parameter
> ("remove=true") to the Record-Route header?  If Remove-Route will be
> present in the general headers, why the "remove=true" parameter?

Email edit error. I initially wrote it as part of Record-Route, and then
changed my mind, but forgot to remote ;remove=true. So, its a new header
(I'm not proposing it yet, though.... I'm still looking for some cases
where its needed).

> 
> But yes, this should work, although deployed proxies may still think
> that if they have included "Record-Route" in a previous request, that's
> all they have to do.

Right, and it is. They will receive all requests for the call. Only if
the client crashes and reboots, then it won't get any subsequent
signaling if recovery takes place. Only new proxies would. Also, only
new proxies can remove themselves. But thats all fine - new features in
the new proxies, but still backwards compatible.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Tue May 30 00:52:08 2000
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Vijay K Gurbani wrote:
> 
> > In addition to this, we can define an explicit "remove me" record-route
> > entry. Something like:
> >
> > Remove-Route: <sip:user@host;maddr=1.2.3.4>;remove=true
> >
> > Proxies would add this when they want out. A UA not understanding this
> > will ignore out continue to use the old route, but a new UA would know
> > that this means to remove it from the Route in the next request. I think
> > this is also backwards compatible.
> >
> > Both of these mechanisms combined (remove-route and adding record-route
> > for mid-call messages) has the desired result of being backwards
> > compatible, providing for restarts after crashes, and allowing proxies
> > to remove themselves during a call.
> 
> Jonathan:
> 
> Are you proposing a new header called "Remove-Route" or a new parameter
> ("remove=true") to the Record-Route header?  If Remove-Route will be
> present in the general headers, why the "remove=true" parameter?

Email edit error. I initially wrote it as part of Record-Route, and then
changed my mind, but forgot to remote ;remove=true. So, its a new header
(I'm not proposing it yet, though.... I'm still looking for some cases
where its needed).

> 
> But yes, this should work, although deployed proxies may still think
> that if they have included "Record-Route" in a previous request, that's
> all they have to do.

Right, and it is. They will receive all requests for the call. Only if
the client crashes and reboots, then it won't get any subsequent
signaling if recovery takes place. Only new proxies would. Also, only
new proxies can remove themselves. But thats all fine - new features in
the new proxies, but still backwards compatible.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Tue May 30 02:38:42 2000
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Hi all,

I am about to start working on the open source SIP code.

Can anybody tell me, which all OSes are supported by the code except LINUX?

I had an overview of the code and could find some lines with "#ifdef
VXWORKS".
Has this code been ported to VXWORKS?

Thanks & Regards,
Nishith.

____________________________________________________________________
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From sip-admin@lists.bell-labs.com  Tue May 30 03:35:50 2000
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Nishith,

Which open source SIP code?  I am aware of a few projects.

Paul

----- Original Message -----
From: "Nishith Chudasama" <nishith76@usa.net>
To: <sip@lists.bell-labs.com>
Sent: Tuesday, May 30, 2000 2:38 AM
Subject: [SIP] OS support


> Hi all,
>
> I am about to start working on the open source SIP code.
>
> Can anybody tell me, which all OSes are supported by the code except
LINUX?
>
> I had an overview of the code and could find some lines with "#ifdef
> VXWORKS".
> Has this code been ported to VXWORKS?
>
> Thanks & Regards,
> Nishith.
>
> ____________________________________________________________________
> Get free email and a permanent address at http://www.netaddress.com/?N=1
>
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>



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From sip-admin@lists.bell-labs.com  Tue May 30 04:44:36 2000
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Gentlemen,

> > > Presumably, this function (TCP/UDP translation) would be provided
> > > only if the clients couldn't talk to each other directly (e.g.
> > > one is TCP only, and the other is UDP only). In this case, the
> > > proxy must remain in the path to continue to provide TCP/UDP
> > > translation for the duration of the call leg.
> > 
> > Since then, some additional discussion seems to indicate that things
> > get a lot easier if we require UAs to speak UDP as the lingua franca.
> > Otherwise, we can't satisfy the goal of simply plugging in a bunch of
> > SIP phones and setting up calls, without a proxy.

So are we saying that adding a Record-Route header is not necessary
just because a proxy is acting as a TCP/UDP bridge?

> To be specific, the proposal is that UDP support in UA transition from
> SHOULD to MUST, so that:
> 
> 1. UAs SHOULD support TCP, MUST support UDP
> 2. proxies MUST support TCP and UDP
> 
> The transport parameter in the Contact and Record-Route headers then
> becomes the preferential choice; if not supported, you can always fall
> back to UDP.

Hmmm... isn't the implication of this that servers should ensure
that for every TCP port listened on (apologies for the
socket-oriented terminology), they have also bound a UDP socket
with the same port number?  Furthermore, these ports should have
the same semantics?  (I can imagine a proxy that exhibits different
behaviour on different ports.)

Is this ever going to be a problem?  (I suspect not, but it prob.
needs to be stated, if not already.)


 - Jo.



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From sip-admin@lists.bell-labs.com  Tue May 30 05:14:37 2000
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Subject: RE: [SIP] wg last call on reliable provisional responses
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Hi,

(SIP List: Apologies for the use of "Gentlemen" in my last post;
 very poor indeed.)

> I've just submitted an updated version of the I-D on "Reliability of
> Provisional Responses in SIP". Until it appears in the archives, you
> can find a copy at:
[...]

Well it's certainly easier for a proxy than the original draft (00).
Which makes me happy. &:)

One thing I'm a little confused about: Section 5.2 (UAS Behavior)
contains the following text:
   "Note that a UAS can send reliable provisional responses
    for any request, including a PRACK request. It is anticipated
    that reliable provisional responses will be most useful for
    INVITE requests."
Is allowing a PRACK to be responded to with reliable provisional
responses potentially a big old nightmare?  Or am I just being a
little bit dull, really?  Also, shouldn't it be stated that
reliable provisional responses can't be used with ACK?

Cheers,


 - Jo.







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From sip-admin@lists.bell-labs.com  Tue May 30 05:27:19 2000
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Jonathan Rosenberg wrote:

> Samandi Sami wrote:
> >
> > Hello,
> >
> > Is there any document (or work) out there describing the distribution of
> > logic between a feature server (or application server) and the underlying
> > network server asking for the services.

The Softswitch consortium's Application Working Group are interested in this area
you might want to also look into their work ...
         http://www.softswitch.org/Working_Groups/application.html

> It is true this is an implementation
> > issue but it is also true that this split of the roles between a SIP
> > application server and a SIP proxy server is similar to the IN architecture
> > where the SSP handles the call control and the SCP the service control.
>
> The difference between and applications server and a proxy is only in
> the complexity of the programs which guide their operation.
>
> Both speak SIP; both can proxy or redirect; there is no way to
> distinguish a proxy from an applications server from the outside. This
> is quite different from the SSP/SCP model, which presumes totally
> different interfaces and roles.

I am not part of the Softswitch Application Working Group, and certainly do not
speak for them, but to me this doesn't appear to totally fit with what I
understand to be their position. I know there are people on this list close to
this work, perhaps they would care to comment.

The Softswitch Application Group are seeking to define the interfaces between an
Application Server and a Softswitch. As I understand it a SIP Proxy Server can be
one example of a Softswitch. They have described the Softswitch as providing basic
call control and signalling, managing resources, and generating call detail
records. Whereas application servers host the service logic for a variety of
enhanced services and media servers provide bearer oriented functions, such as
IVR, conferencing, and fax. They suggest SIP is used as the interface between an
Application Server and a Softswitch, while the Real Time Protocol (RTP) is used as
the interface between a media server and media gateway.

Please note I am not seeking to promote the Softswitch view over the one described
by Jonathan, which I personally empathise with, just some discussion of a very
interesting area.

Regards,
Neil
--
Ubiquity Software Corporation, UK           http://www.ubiquity.net

> I think the SIP model is a very powerful
> model. It means that decisions about service invocation can always be
> represented as routing decisions. If I want some service from provider
> X, all that needs to happen is to have my call routed to the server at
> domain X.
>
> One of the reasons the separation between call control and service
> control was made in the IN, was the desire for the SCP not to receive
> all signaling messages for the call. I will note that in SIP, this
> function is already provided in the call signaling itself, through the
> Record-Route mechanism. This means an application server can drop out of
> the call after routing the initial request, as can a proxy.
>
>  Is
> > it then logical to think about an implementation like the following :
> >
> > -       The network server is a call stateful proxy (holding the state of a
> > call between the INVITE and the BYE). This SIP proxy forward requests to the
> > application server. This proxy server goes stateful and proxy requests only
> > when needed (for instance when the end user has subscribed to predefined
> > services).
>
> Why? As I argue above, this distinction of an SSP/SCP role makes less
> sense in SIP.
>
> > -       The application server will behave per call basis whether as a
> > redirect server (to provide routing services) or as a UAS when media
> > services are needed (IVR, ...). I am not sure whether this equipment should
> > also proxy the request ?.
>
> If that is part of the service, sure.
>
> > -       The goal here is not to involve the application server for the whole
> > duration of the call but only when needed and this means telling the
> > application server what's state of the call that triggered the call of the
> > service. For instance, A call B, but  B is busy,  the Proxy server of B
> > receives a 486 and then sends an INVITE to the feature server asking what to
> > do. How to tell the application server that the user is busy (the equivalent
> > of the DP Terminal busy of the IN) ? Should this be done in a proprietary
> > way ?
>
> I would not do it this way. I would send the initial INVITE to the
> feature server, and when the 486 response comes, have it go from there.



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From sip-admin@lists.bell-labs.com  Tue May 30 07:37:44 2000
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From: Lars Berggren <lars@intertex.se>
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I quote:
"We define the calling URI as the URI found in the Contact header of the
 request from the calling UA, if present, or the From header, if there is
 no Contact header. The called UA copies the Record-Route entries,
 maintaining their ordering, to the Route header field. However, all
 components except the maddr parameter are replaced by the
 calling URI."

As I interpret this, a proxy can not specify which port or the transport
protocol it wants to be used in subsequent requests from the called UA 
since the only parameter of the url of the proxy that is reflected in the
Route-list is maddr. Is this correct? 

What about rr-extension, is it supposed to be copied into the route field 
as a route-extension or is that supposed to be defined by the extension?

I think it could be useful for a proxy to signal that it wants to have the
rr-extension copied into subsequent requests in some way. For example, a
call stateful proxy could store call state in a rr-extension parameter.

Thanks

/Lars


Lars Berggren       <mailto:lars.berggren@intertex.se>
Intertex Data AB    tel: +46-8-6282828
Sundbyberg, Sweden  fax: +46-8-6286414



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Organization: Oy L M Ericsson Ab
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Hi,

In section 10.5.1 of the bis document it is said that when UDP is used a
final response is retransmitted until:

1) ACK arrives
2) BYE arrives
3) CANCEL arrives (status code > 299)
4) 7 retransmissions

In section 4.2.5 it is said that CANCEL has to be answered with 487.

So, if a server receives an INVITE and it responses with 400 for
instance. If a CANCEL is received, it will stop sending the 400 and it
will send the 487. It will retransmit the 487 until the ACK for the
INVITE arrives. Does the retransmission counter get re-started (the
server might have retransmitted already some times the 400 response)?

Thanks,

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


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Jonathan Rosenberg wrote:
> 
> Plong@smithmicro.com wrote:
> >
> > Hello, all. Just starting with SIP, but I already have a few basic questions
> > regarding the RFC. Since this is my first query, let me know if these
> > questions should be posted elsewhere or I have otherwise violated some other
> > rule of etiquette.
> 
> Many common questions are answered in the FAQ:
> http://www.cs.columbia.edu/~hgs/sip/faq.html
> 
> >
> > - "...SIP...is...for...sessions with one or more participants." - How can
> > there be a session with only one participant? A session is defined elsewhere
> > as comprising one or more RTP sessions, which implies a sender and a
> > receiver, or two participants. Should the sentence have been written as
> > "...with two or more participants?"
> 
> Mulicast sessions can have a single participant. Not that interesting,
> but possible.
> 

Note that a "session" does not have to be RTP. It might have an RTP
component, might have several RTP components, or none at all. I might
also have an associated "Quake" game, SIGTRAN connection, or anything
else that can be described  from a third-person perspective.

> > - "Servers are either proxy, redirect or user agent servers or
> > registrars." - I have also seen references to "location servers" and "SIP
> > servers." Is a location server the same thing as a redirect server? Is "SIP
> > server" just a general reference to servers in "SIPdom?"
> 
> Location server is a non-SIP server, like an LDAP server or SQL
> database. "SIP server" is any entity that receives requests (UAS, proxy,
> redirect, registrar).

This may be a slightly misleading statements. Many implemented Location
Servers are in fact SIP Servers hat have access to additional routing
data such as LDAP.

--
Dean


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Hi Gonzalo,
     Answers inline.




Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se> on 05/30/2000
05:34:06 PM

To:   sip <sip@lists.research.bell-labs.com>
cc:    (bcc: K Binu/HSSBLR)

Subject:  [SIP] Retransmissions




>Hi,
>
>In section 10.5.1 of the bis document it is said that when UDP is used a
>final response is retransmitted until:
>
>1) ACK arrives
>2) BYE arrives
>3) CANCEL arrives (status code > 299)
>4) 7 retransmissions
>
>In section 4.2.5 it is said that CANCEL has to be answered with 487.
>

The section does not say that CANCEL is to be answered with a 487. The
CANCEL has to be answered with a 200 OK. The 487 response is to be sent if
a retransmission of the cancelled INVITE request comes by.

>So, if a server receives an INVITE and it responses with 400 for
>instance. If a CANCEL is received, it will stop sending the 400 and it
>will send the 487. It will retransmit the 487 until the ACK for the
>INVITE arrives. Does the retransmission counter get re-started (the
>server might have retransmitted already some times the 400 response)?

So the correct behaviour here would be to stop the 400 response
retransmissions for the INVITE and answer the CANCEL with a 200 OK. The
server would send a 487 if the cancelled INVITE happens to come by and not
otherwise.

>
>Thanks,
>
>Gonzalo
>--
>Gonzalo Camarillo         Phone :  +358  9 299 33 71
>Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
>Telecom R&D               Fax   :  +358  9 299 30 52
>FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
>Finland
>
>
>______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip

Binu




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From sip-admin@lists.bell-labs.com  Tue May 30 09:00:46 2000
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hi,

I'm querying the use of the hidden parameter and received parameter in the same via.

It is the purpose of the hidden parameter to hide the previous hop from subsequent hopes and this 
is done by encrypting the 'sent-by' part of the Via header.

However, it is possible that the Proxy has added a received parameter to the encrypted via in order
to reply to the correct machine.  This received parameter provides open knowledge of the previous 
hop's address and compromises the security.

Now there are two possibilities here (as far as i can see):

1) The received parameter should not be added to the hidden via but cached by the proxy in a lookup
    table whilst it waits for the response. (or the received parameter can be encrypted as well).

2) For a received parameter to be inserted, then the message has probably come through a NAT or firewall.
    this is sufficent enough for security matters and aslong as the original host is encrypted within the via
   i.e. the sent-by remains encrypted, all is OK.

any thoughts?

-- 
Gethin Liddell
Ubiquity Software Corporation

http://www.ubiquity.net
mailto:gethin@ubiquity.net


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Can somebody please answer the following question? "

     How can you have a "session" of any kind with only one participant?

charles
Dean Willis wrote:

> Jonathan Rosenberg wrote:
> >
> > Plong@smithmicro.com wrote:
> > >
> > > Hello, all. Just starting with SIP, but I already have a few basic questions
> > > regarding the RFC. Since this is my first query, let me know if these
> > > questions should be posted elsewhere or I have otherwise violated some other
> > > rule of etiquette.
> >
> > Many common questions are answered in the FAQ:
> > http://www.cs.columbia.edu/~hgs/sip/faq.html
> >
> > >
> > > - "...SIP...is...for...sessions with one or more participants." - How can
> > > there be a session with only one participant? A session is defined elsewhere
> > > as comprising one or more RTP sessions, which implies a sender and a
> > > receiver, or two participants. Should the sentence have been written as
> > > "...with two or more participants?"
> >
> > Mulicast sessions can have a single participant. Not that interesting,
> > but possible.
> >
>
> Note that a "session" does not have to be RTP. It might have an RTP
> component, might have several RTP components, or none at all. I might
> also have an associated "Quake" game, SIGTRAN connection, or anything
> else that can be described  from a third-person perspective.
>
> > > - "Servers are either proxy, redirect or user agent servers or
> > > registrars." - I have also seen references to "location servers" and "SIP
> > > servers." Is a location server the same thing as a redirect server? Is "SIP
> > > server" just a general reference to servers in "SIPdom?"
> >
> > Location server is a non-SIP server, like an LDAP server or SQL
> > database. "SIP server" is any entity that receives requests (UAS, proxy,
> > redirect, registrar).
>
> This may be a slightly misleading statements. Many implemented Location
> Servers are in fact SIP Servers hat have access to additional routing
> data such as LDAP.
>
> --
> Dean
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--

Charles Agboh

Universite Libre de Bruxelles
Service Telematique et Communication
Bd du Triomphe, CP 230
B-1050 Brussels
Belgium

Tel. +32 2 6505717
Fax. +32 2 6505088, +32 2 6293816
RFC-822 : agboh@helios.iihe.ac.be

----------------------------------
"Perilous to us all are the devices of an art deeper than we possess ourselves."
--J.R.R Tolkien, The Two Towers




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From sip-admin@lists.bell-labs.com  Tue May 30 09:26:27 2000
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I know my head is shaped like a bell, but a self-test on a given interface
is rather common.

-----Original Message-----
From: charles Agboh [mailto:agboh@helios.iihe.ac.be]
Sent: Tuesday, May 30, 2000 8:06 AM
To: Dean Willis; sip@lists.bell-labs.com
Cc: Jonathan Rosenberg
Subject: Re: [SIP] Basic questions



Can somebody please answer the following question? "

     How can you have a "session" of any kind with only one participant?

charles
Dean Willis wrote:

> Jonathan Rosenberg wrote:
> >
> > Plong@smithmicro.com wrote:
> > >
> > > Hello, all. Just starting with SIP, but I already have a few basic
questions
> > > regarding the RFC. Since this is my first query, let me know if these
> > > questions should be posted elsewhere or I have otherwise violated some
other
> > > rule of etiquette.
> >
> > Many common questions are answered in the FAQ:
> > http://www.cs.columbia.edu/~hgs/sip/faq.html
> >
> > >
> > > - "...SIP...is...for...sessions with one or more participants." - How
can
> > > there be a session with only one participant? A session is defined
elsewhere
> > > as comprising one or more RTP sessions, which implies a sender and a
> > > receiver, or two participants. Should the sentence have been written
as
> > > "...with two or more participants?"
> >
> > Mulicast sessions can have a single participant. Not that interesting,
> > but possible.
> >
>
> Note that a "session" does not have to be RTP. It might have an RTP
> component, might have several RTP components, or none at all. I might
> also have an associated "Quake" game, SIGTRAN connection, or anything
> else that can be described  from a third-person perspective.
>
> > > - "Servers are either proxy, redirect or user agent servers or
> > > registrars." - I have also seen references to "location servers" and
"SIP
> > > servers." Is a location server the same thing as a redirect server? Is
"SIP
> > > server" just a general reference to servers in "SIPdom?"
> >
> > Location server is a non-SIP server, like an LDAP server or SQL
> > database. "SIP server" is any entity that receives requests (UAS, proxy,
> > redirect, registrar).
>
> This may be a slightly misleading statements. Many implemented Location
> Servers are in fact SIP Servers hat have access to additional routing
> data such as LDAP.
>
> --
> Dean
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--

Charles Agboh

Universite Libre de Bruxelles
Service Telematique et Communication
Bd du Triomphe, CP 230
B-1050 Brussels
Belgium

Tel. +32 2 6505717
Fax. +32 2 6505088, +32 2 6293816
RFC-822 : agboh@helios.iihe.ac.be

----------------------------------
"Perilous to us all are the devices of an art deeper than we possess
ourselves."
--J.R.R Tolkien, The Two Towers




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From sip-admin@lists.bell-labs.com  Tue May 30 09:33:00 2000
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Date: Tue, 30 May 2000 15:34:40 +0200 (CEST)
From: Lars Berggren <lars@intertex.se>
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On Tue, 30 May 2000, charles Agboh wrote:

> 
> Can somebody please answer the following question? "
> 
>      How can you have a "session" of any kind with only one participant?
> 
> charles

Jonathan gave a good example: multicast sessions. Imagine you are the only
one which sends and/or receives "session data" to a multicast ip-address.
Then you might end up "talking to yourself", but still, it *is* a session
with only one participant.

/Lars

Lars Berggren       <mailto:lars.berggren@intertex.se>
Intertex Data AB    tel: +46-8-6282828
Sundbyberg, Sweden  fax: +46-8-6286414



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From sip-admin@lists.bell-labs.com  Tue May 30 09:36:38 2000
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Subject: RE: [SIP] SIP Application Server
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The softswitch consortium applications working group is working on an
architecture supporting enhanced services in the softswitch network
architecture. An application server is introduced into the architecture to
host enhanced service logic and management functions. While a softswitch
provides basic call control and signaling, manages resources, and generates
call detail records, application servers are intended to host a variety of
enhanced services. Examples of enhanced services include calling card, 800
number translation, Centrex, voice mail, and find-me services. SIP is used
as the interface between an application server and a softswitch. It allows a
softswitch to direct calls to an application server for enhanced services
processing and also allows an application server to redirect or transfer
calls back through a softswitch.

An applications server can be a proxy, redirect server or user agent. It may
be stateful or stateless. It really depends on the applications being
provided. I've seen applications servers which are simple redirect servers
providing routing functions and more complex ones acting as back-to-back
user agents, providing unified messaging. Using SIP allows a softswitch (or
other end user device) to pass control to an application server, without
having to worry about its architecture or functions.

Scott Hoffpauir
BroadSoft, Inc.
200 Perry Parkway, Suite 1
Gaithersburg, MD 20878
301-977-4295

-----Original Message-----
From: sip-admin@lists.bell-labs.com
[mailto:sip-admin@lists.bell-labs.com]On Behalf Of Neil Deason
Sent: Tuesday, May 30, 2000 5:25 AM
To: sip@lists.bell-labs.com
Subject: Re: [SIP] SIP Application Server


Jonathan Rosenberg wrote:

> Samandi Sami wrote:
> >
> > Hello,
> >
> > Is there any document (or work) out there describing the distribution of
> > logic between a feature server (or application server) and the
underlying
> > network server asking for the services.

The Softswitch consortium's Application Working Group are interested in this
area
you might want to also look into their work ...
         http://www.softswitch.org/Working_Groups/application.html

> It is true this is an implementation
> > issue but it is also true that this split of the roles between a SIP
> > application server and a SIP proxy server is similar to the IN
architecture
> > where the SSP handles the call control and the SCP the service control.
>
> The difference between and applications server and a proxy is only in
> the complexity of the programs which guide their operation.
>
> Both speak SIP; both can proxy or redirect; there is no way to
> distinguish a proxy from an applications server from the outside. This
> is quite different from the SSP/SCP model, which presumes totally
> different interfaces and roles.

I am not part of the Softswitch Application Working Group, and certainly do
not
speak for them, but to me this doesn't appear to totally fit with what I
understand to be their position. I know there are people on this list close
to
this work, perhaps they would care to comment.

The Softswitch Application Group are seeking to define the interfaces
between an
Application Server and a Softswitch. As I understand it a SIP Proxy Server
can be
one example of a Softswitch. They have described the Softswitch as providing
basic
call control and signalling, managing resources, and generating call detail
records. Whereas application servers host the service logic for a variety of
enhanced services and media servers provide bearer oriented functions, such
as
IVR, conferencing, and fax. They suggest SIP is used as the interface
between an
Application Server and a Softswitch, while the Real Time Protocol (RTP) is
used as
the interface between a media server and media gateway.

Please note I am not seeking to promote the Softswitch view over the one
described
by Jonathan, which I personally empathise with, just some discussion of a
very
interesting area.

Regards,
Neil
--
Ubiquity Software Corporation, UK           http://www.ubiquity.net

> I think the SIP model is a very powerful
> model. It means that decisions about service invocation can always be
> represented as routing decisions. If I want some service from provider
> X, all that needs to happen is to have my call routed to the server at
> domain X.
>
> One of the reasons the separation between call control and service
> control was made in the IN, was the desire for the SCP not to receive
> all signaling messages for the call. I will note that in SIP, this
> function is already provided in the call signaling itself, through the
> Record-Route mechanism. This means an application server can drop out of
> the call after routing the initial request, as can a proxy.
>
>  Is
> > it then logical to think about an implementation like the following :
> >
> > -       The network server is a call stateful proxy (holding the state
of a
> > call between the INVITE and the BYE). This SIP proxy forward requests to
the
> > application server. This proxy server goes stateful and proxy requests
only
> > when needed (for instance when the end user has subscribed to predefined
> > services).
>
> Why? As I argue above, this distinction of an SSP/SCP role makes less
> sense in SIP.
>
> > -       The application server will behave per call basis whether as a
> > redirect server (to provide routing services) or as a UAS when media
> > services are needed (IVR, ...). I am not sure whether this equipment
should
> > also proxy the request ?.
>
> If that is part of the service, sure.
>
> > -       The goal here is not to involve the application server for the
whole
> > duration of the call but only when needed and this means telling the
> > application server what's state of the call that triggered the call of
the
> > service. For instance, A call B, but  B is busy,  the Proxy server of B
> > receives a 486 and then sends an INVITE to the feature server asking
what to
> > do. How to tell the application server that the user is busy (the
equivalent
> > of the DP Terminal busy of the IN) ? Should this be done in a
proprietary
> > way ?
>
> I would not do it this way. I would send the initial INVITE to the
> feature server, and when the 486 response comes, have it go from there.



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Date: Tue, 30 May 2000 13:37:39 +0000
From: "Sinnreich, Henry" <Henry.Sinnreich@WCOM.Com>
Subject: RE: [SIP] OS support
To: "'Nishith Chudasama'" <nishith76@usa.net>, sip@lists.bell-labs.com
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See an extensive list at
http://www.cs.columbia.edu/sip/implementations.html

and other several sources, such as

http://www.vovida.com/content.cfm?pagename=products

Henry

Henry Sinnreich
MCI WorldCom
400 International Parkway
Richardson, Texas 75081
USA 

>-----Original Message-----
>From: Nishith Chudasama [mailto:nishith76@usa.net]
>Sent: Tuesday, May 30, 2000 1:39 AM
>To: sip@lists.bell-labs.com
>Subject: [SIP] OS support
>
>
>Hi all,
>
>I am about to start working on the open source SIP code.
>
>Can anybody tell me, which all OSes are supported by the code 
>except LINUX?
>
>I had an overview of the code and could find some lines with "#ifdef
>VXWORKS".
>Has this code been ported to VXWORKS?
>
>Thanks & Regards,
>Nishith.
>
>____________________________________________________________________
>Get free email and a permanent address at 
>http://www.netaddress.com/?N=1
>
>
>_______________________________________________
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>http://lists.bell-labs.com/mailman/listinfo/sip
>


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From sip-admin@lists.bell-labs.com  Tue May 30 09:41:56 2000
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From: Dean Willis <dwillis@dynamicsoft.com>
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I regularly multicast myself to no particular audience. Occasionally I
join the multicast to see if I'm still there and tell people I'm
checking up on myself. Lacking WAN mulicast routing, my traffic doesn't
go all that far -- but it's still a session, even when nobody is joined.

Hence "one or more" participants rather than "zero or more".

--
Dean

charles Agboh wrote:
> 
> Can somebody please answer the following question? "
> 
>      How can you have a "session" of any kind with only one participant?
> 
> charles
> Dean Willis wrote:
> 
> > Jonathan Rosenberg wrote:
> > >
> > > Plong@smithmicro.com wrote:
> > > >
> > > > Hello, all. Just starting with SIP, but I already have a few basic questions
> > > > regarding the RFC. Since this is my first query, let me know if these
> > > > questions should be posted elsewhere or I have otherwise violated some other
> > > > rule of etiquette.
> > >
> > > Many common questions are answered in the FAQ:
> > > http://www.cs.columbia.edu/~hgs/sip/faq.html
> > >
> > > >
> > > > - "...SIP...is...for...sessions with one or more participants." - How can
> > > > there be a session with only one participant? A session is defined elsewhere
> > > > as comprising one or more RTP sessions, which implies a sender and a
> > > > receiver, or two participants. Should the sentence have been written as
> > > > "...with two or more participants?"
> > >
> > > Mulicast sessions can have a single participant. Not that interesting,
> > > but possible.
> > >
> >
> > Note that a "session" does not have to be RTP. It might have an RTP
> > component, might have several RTP components, or none at all. I might
> > also have an associated "Quake" game, SIGTRAN connection, or anything
> > else that can be described  from a third-person perspective.
> >
> > > > - "Servers are either proxy, redirect or user agent servers or
> > > > registrars." - I have also seen references to "location servers" and "SIP
> > > > servers." Is a location server the same thing as a redirect server? Is "SIP
> > > > server" just a general reference to servers in "SIPdom?"
> > >
> > > Location server is a non-SIP server, like an LDAP server or SQL
> > > database. "SIP server" is any entity that receives requests (UAS, proxy,
> > > redirect, registrar).
> >
> > This may be a slightly misleading statements. Many implemented Location
> > Servers are in fact SIP Servers hat have access to additional routing
> > data such as LDAP.
> >
> > --
> > Dean
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> --
> 
> Charles Agboh
> 
> Universite Libre de Bruxelles
> Service Telematique et Communication
> Bd du Triomphe, CP 230
> B-1050 Brussels
> Belgium
> 
> Tel. +32 2 6505717
> Fax. +32 2 6505088, +32 2 6293816
> RFC-822 : agboh@helios.iihe.ac.be
> 
> ----------------------------------
> "Perilous to us all are the devices of an art deeper than we possess ourselves."
> --J.R.R Tolkien, The Two Towers


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Dean,

So is this statement incorrect? Section 1.3: "A session as defined for SDP
can comprise one or more RTP sessions."

Paul Long (an admitted hair splitter)
Smith Micro Software, Inc.


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From sip-admin@lists.bell-labs.com  Tue May 30 12:41:40 2000
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Thanks for providing the authoritive voice from the working group Scott.

Scott Hoffpauir wrote:

> The softswitch consortium applications working group is working on an
> architecture supporting enhanced services in the softswitch network
> architecture. An application server is introduced into the architecture to
> host enhanced service logic and management functions. While a softswitch
> provides basic call control and signaling, manages resources, and generates
> call detail records, application servers are intended to host a variety of
> enhanced services. Examples of enhanced services include calling card, 800
> number translation, Centrex, voice mail, and find-me services. SIP is used
> as the interface between an application server and a softswitch. It allows a
> softswitch to direct calls to an application server for enhanced services
> processing and also allows an application server to redirect or transfer
> calls back through a softswitch.
>
> An applications server can be a proxy, redirect server or user agent. It may
> be stateful or stateless. It really depends on the applications being
> provided. I've seen applications servers which are simple redirect servers
> providing routing functions and more complex ones acting as back-to-back
> user agents, providing unified messaging. Using SIP allows a softswitch (or
> other end user device) to pass control to an application server, without
> having to worry about its architecture or functions.

My apologies if I am confusing the terms here but am I wrong in thinking that a
SIP Proxy Server qualifies as a Softswitch? I have seen them described as such
within the Softswitch Consortium. If a SIP Proxy is a Softswitch, isn't this
approach in danger of reproducing the SSP/SCP model? Why would a Proxy Server
need to pass an INVITE on to an Application Server for the AS to then
proxy/redirect it? Couldn't SIP-CGI and CPL provide ways for a Proxy to access
any service logic required to decide whether to proxy/redirect it directly. A
Proxy Server augmented with service logic seems to be both a Softswitch and an
Application Server, which would make defining an interface based on SIP between
the two interesting.

Regards,
Neil.
--
Ubiquity Software Corporation, UK          http://www.ubiquity.net

> Jonathan Rosenberg wrote:
>
> > Samandi Sami wrote:
> > >
> > > Hello,
> > >
> > > Is there any document (or work) out there describing the distribution of
> > > logic between a feature server (or application server) and the
> underlying
> > > network server asking for the services.
>
> The Softswitch consortium's Application Working Group are interested in this
> area
> you might want to also look into their work ...
>          http://www.softswitch.org/Working_Groups/application.html
>
> > It is true this is an implementation
> > > issue but it is also true that this split of the roles between a SIP
> > > application server and a SIP proxy server is similar to the IN
> architecture
> > > where the SSP handles the call control and the SCP the service control.
> >
> > The difference between and applications server and a proxy is only in
> > the complexity of the programs which guide their operation.
> >
> > Both speak SIP; both can proxy or redirect; there is no way to
> > distinguish a proxy from an applications server from the outside. This
> > is quite different from the SSP/SCP model, which presumes totally
> > different interfaces and roles.
>
> I am not part of the Softswitch Application Working Group, and certainly do
> not
> speak for them, but to me this doesn't appear to totally fit with what I
> understand to be their position. I know there are people on this list close
> to
> this work, perhaps they would care to comment.
>
> The Softswitch Application Group are seeking to define the interfaces
> between an
> Application Server and a Softswitch. As I understand it a SIP Proxy Server
> can be
> one example of a Softswitch. They have described the Softswitch as providing
> basic
> call control and signalling, managing resources, and generating call detail
> records. Whereas application servers host the service logic for a variety of
> enhanced services and media servers provide bearer oriented functions, such
> as
> IVR, conferencing, and fax. They suggest SIP is used as the interface
> between an
> Application Server and a Softswitch, while the Real Time Protocol (RTP) is
> used as
> the interface between a media server and media gateway.
>
> Please note I am not seeking to promote the Softswitch view over the one
> described
> by Jonathan, which I personally empathise with, just some discussion of a
> very
> interesting area.
>
> Regards,
> Neil
> --
> Ubiquity Software Corporation, UK          http://www.ubiquity.net
>
> > I think the SIP model is a very powerful
> > model. It means that decisions about service invocation can always be
> > represented as routing decisions. If I want some service from provider
> > X, all that needs to happen is to have my call routed to the server at
> > domain X.
> >
> > One of the reasons the separation between call control and service
> > control was made in the IN, was the desire for the SCP not to receive
> > all signaling messages for the call. I will note that in SIP, this
> > function is already provided in the call signaling itself, through the
> > Record-Route mechanism. This means an application server can drop out of
> > the call after routing the initial request, as can a proxy.
> >
> >  Is
> > > it then logical to think about an implementation like the following :
> > >
> > > -       The network server is a call stateful proxy (holding the state
> of a
> > > call between the INVITE and the BYE). This SIP proxy forward requests to
> the
> > > application server. This proxy server goes stateful and proxy requests
> only
> > > when needed (for instance when the end user has subscribed to predefined
> > > services).
> >
> > Why? As I argue above, this distinction of an SSP/SCP role makes less
> > sense in SIP.
> >
> > > -       The application server will behave per call basis whether as a
> > > redirect server (to provide routing services) or as a UAS when media
> > > services are needed (IVR, ...). I am not sure whether this equipment
> should
> > > also proxy the request ?.
> >
> > If that is part of the service, sure.
> >
> > > -       The goal here is not to involve the application server for the
> whole
> > > duration of the call but only when needed and this means telling the
> > > application server what's state of the call that triggered the call of
> the
> > > service. For instance, A call B, but  B is busy,  the Proxy server of B
> > > receives a 486 and then sends an INVITE to the feature server asking
> what to
> > > do. How to tell the application server that the user is busy (the
> equivalent
> > > of the DP Terminal busy of the IN) ? Should this be done in a
> proprietary
> > > way ?
> >
> > I would not do it this way. I would send the initial INVITE to the
> > feature server, and when the 486 response comes, have it go from there.



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I propose the following changes to draft-ietf-sip-2543bis-00:

Page 59, BNF of sent-by for Via header should read:

sent-by = ([user@]host[:port])|(concealed-host)

where user is as defined in Section 2, on page 18.

Also, in section 4.2.6, page 28, description of Request-URI, delete the
sentence "The user name MUST be empty."


The reason for both changes is that we have a model in which a SIP Server
is providing proxy services to a large collection of users.  It often
happens that a client belonging to user1 will send a SIP request such as
an INVITE to this server.  The server will receive this request and make
various decisions according to logic installed on behalf of user1.  This
logic frequently requires stateful interactions.  One decision could be to
proxy the INVITE to the server for the invited party (user2).  The proxy
server for user2 is located on the same host (and same port) as the server
for user1.  The server for user2 may create the response to the INVITE or
it may proxy it onwards to one or more clients for user2, or perhaps to
other users that have been designated by user2.  This is all according to
logic installed on behalf of user2. When the response comes back to the
server we want the server for user2 to see the response (and perhaps
decide to proxy the request to SIP entities) before the response is seen
by the server for user1.  The addition of the user part in the sent-by of
the via allows such a routing.  If it is ever used in a situation where
there is no concept of different servers for different users it is
ignored.

Note that the branch parameter does not meet our needs since there may, in
fact, be no branching.  Also, note that a common port on the SIP Server
services all user agents, so the port parameter could not be used to
differentiate servers for different users.  And, there may be more users
serviced on this host than could be separated with a typical 16 bit port
number.

The suggestion for removing the sentence about the user part in the
Request-URI of REGISTER requests is similarly motivated. There is no
reason given in the rfc why the user part MUST be empty in a Request-URI
for REGISTER.  In our model, a client will want to REGISTER itself with
the server representing it within the SIP Server.  In practice, it would
be sufficient to use the From or To field of the REGISTER to determine
which server should be used, but in a large system, where the servers
might be distributed among many hosts (even though they are seen as a
single SIP server host to the outside world) it is more efficient to use
the Request-URI for routing purposes so we want the user name to be part
of the Request-URI.

Ray Zibman
Senior Scientist
MetaTel, Inc.
45 Rumford Ave.
Waltham, MA 02453
(781) 392-2509





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From sip-admin@lists.bell-labs.com  Tue May 30 13:39:39 2000
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Ray Zibman wrote:

> I propose the following changes to draft-ietf-sip-2543bis-00:
>
> Page 59, BNF of sent-by for Via header should read:
>
> sent-by = ([user@]host[:port])|(concealed-host)
>
> where user is as defined in Section 2, on page 18.
>
> Also, in section 4.2.6, page 28, description of Request-URI, delete the
> sentence "The user name MUST be empty."

Sorry, but I disagree.

> The reason for both changes is that we have a model in which a SIP Server
> is providing proxy services to a large collection of users.  It often
> happens that a client belonging to user1 will send a SIP request such as
> an INVITE to this server.  The server will receive this request and make
> various decisions according to logic installed on behalf of user1.  This
> logic frequently requires stateful interactions.  One decision could be to
> proxy the INVITE to the server for the invited party (user2).  The proxy
> server for user2 is located on the same host (and same port) as the server
> for user1.  The server for user2 may create the response to the INVITE or
> it may proxy it onwards to one or more clients for user2, or perhaps to
> other users that have been designated by user2.  This is all according to
> logic installed on behalf of user2. When the response comes back to the
> server we want the server for user2 to see the response (and perhaps
> decide to proxy the request to SIP entities) before the response is seen
> by the server for user1.  The addition of the user part in the sent-by of
> the via allows such a routing.  If it is ever used in a situation where
> there is no concept of different servers for different users it is
> ignored.

 If I understand you correctly the response will come back to the right place
as the 'two' servers are listening on the same host/port. Not associating the
response with user2 is a local implementation issue. You could for example use
a branch tag, doesn't matter if the request wasn't really forked, or even use
a via-extension param.

> Note that the branch parameter does not meet our needs since there may, in
> fact, be no branching.  Also, note that a common port on the SIP Server
> services all user agents, so the port parameter could not be used to
> differentiate servers for different users.  And, there may be more users
> serviced on this host than could be separated with a typical 16 bit port
> number.
>
> The suggestion for removing the sentence about the user part in the
> Request-URI of REGISTER requests is similarly motivated. There is no
> reason given in the rfc why the user part MUST be empty in a Request-URI
> for REGISTER.  In our model, a client will want to REGISTER itself with
> the server representing it within the SIP Server.  In practice, it would
> be sufficient to use the From or To field of the REGISTER to determine
> which server should be used, but in a large system, where the servers
> might be distributed among many hosts (even though they are seen as a
> single SIP server host to the outside world) it is more efficient to use
> the Request-URI for routing purposes so we want the user name to be part
> of the Request-URI.

There is a reason why the user part is empty, because the Request-URI names
the domain of the registrar, not a user within a domain. If the information
you require already is in the To header I don't think a change is necessary
here either.

hth,
Neil
--
Ubiquity Software Corporation, UK           http://www.ubiquity.net



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From sip-admin@lists.bell-labs.com  Tue May 30 13:50:18 2000
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Does SIP support Call pickup feature, if so where can I find details about
it.

thanks

amar singh
Shoreline Communications Inc.


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From sip-admin@lists.bell-labs.com  Tue May 30 14:00:16 2000
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It can, see p.29 of the bis draft. If a registration is changed while a User
Agent or Proxy Server processes an invitation, the new information SHOULD be
used.

hth,
Neil.
--
Ubiquity Software Corporation, UK           http://www.ubiquity.net

Amar Singh wrote:

> Does SIP support Call pickup feature, if so where can I find details about
> it.
>
> thanks
>
> amar singh
> Shoreline Communications Inc.



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From sip-admin@lists.bell-labs.com  Tue May 30 14:01:47 2000
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Hi,
   does SIP have Domain or Zone concept ?
I would assume it does, cause I saw a Draft describing SIP interdomain
architecture.
If it does, how does it define a Domain or Zone ? by Register on a SIP
Server ?


By the way, is there a API for SIP network?  I saw a Draft Java Servlet API
but that's about it.


Thank you for the time.

Tricia 
NeTrue Communications Inc.



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From: "Sinnreich, Henry" <Henry.Sinnreich@WCOM.Com>
Subject: RE: [SIP] Domain or Zone concept
To: "'Tricia Wang'" <TWang@NeTrue.com>, sip@lists.bell-labs.com
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>I saw a Draft describing SIP interdomain

The draft <sinnreich-sip-qos-osp> assumes that a SIP proxy server within a
domain controls the access to SIP services in that domain. The SIP proxy
also controls access to QoS support by interacting with the policy server
for that domain, to install policy in the network to support QoS for SIP
calls. There is no notion of a "SIP Domain", since SIP services are only a
subset of many other services in the same domain. The domain owns access to
services and not vice versa.

Henry

>-----Original Message-----
>From: Tricia Wang [mailto:TWang@NeTrue.com]
>Sent: Tuesday, May 30, 2000 1:02 PM
>To: sip@lists.bell-labs.com
>Subject: [SIP] Domain or Zone concept
>
>
>Hi,
>   does SIP have Domain or Zone concept ?
>I would assume it does, cause I saw a Draft describing SIP interdomain
>architecture.
>If it does, how does it define a Domain or Zone ? by Register on a SIP
>Server ?
>
>
>By the way, is there a API for SIP network?  I saw a Draft 
>Java Servlet API
>but that's about it.
>
>
>Thank you for the time.
>
>Tricia 
>NeTrue Communications Inc.
>
>
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip
>


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From sip-admin@lists.bell-labs.com  Tue May 30 14:20:58 2000
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Still waiting for my ride home so I'll continue to try and give Jonathan a rest
from having to answer everything...

Tricia Wang wrote:

> Hi,
>    does SIP have Domain or Zone concept ?
> I would assume it does, cause I saw a Draft describing SIP interdomain
> architecture.
> If it does, how does it define a Domain or Zone ? by Register on a SIP
> Server ?

SIP domains are encapsulated within SIP URLs, e.g sip:user@domain. To find a
sip server for a user you might do DNS query for a SIP Server associated with
their domain.  The Request-URI of a REGISTER also names the domain of a
registration request.

> By the way, is there a API for SIP network?  I saw a Draft Java Servlet API
> but that's about it.

Not sure what exactly you mean by an API to a 'SIP network' but there are
various contenders for service creation in a SIP network - CPL, Servlets
SIP-CGI for example. Have a look around Henning's site, in particular ..

    http://www.cs.columbia.edu/%7Ehgs/sip/drafts_api.html

There is also talk of JAIN/PARLAY interfaces to SIP exisiting in the future.
hth,
Neil
--
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> -----Original Message-----
> From: sip-admin@lists.bell-labs.com
> [mailto:sip-admin@lists.bell-labs.com]On Behalf Of charles Agboh
> Sent: Tuesday, May 30, 2000 9:06 AM
> To: Dean Willis; sip@lists.bell-labs.com
> Cc: Jonathan Rosenberg
> Subject: Re: [SIP] Basic questions
> 
> 
> 
> Can somebody please answer the following question? "
> 
>      How can you have a "session" of any kind with only one participant?
> 
> charles

What is the sound of one hand clapping?

SIP can set up Zen sessions. It's completely general.


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Tricia Wang wrote:
> 
> Hi,
>    does SIP have Domain or Zone concept ?
> I would assume it does, cause I saw a Draft describing SIP interdomain
> architecture.
> If it does, how does it define a Domain or Zone ? by Register on a SIP
> Server ?

I would say that the only concept of domains in vanilla SIP is really
those of DNS. At a different level people talk about a particular
providers network consituting a domain of its own, especially when
discussing trust issues. Trust domains will obviously not generally map
onto DNS domains one-to-one.

> 
> By the way, is there a API for SIP network?  I saw a Draft Java Servlet API
> but that's about it.

Right. That would be

http://www.ietf.org/internet-drafts/draft-kristensen-sip-servlet-00.txt

There will hopefully be an update of that draft sometime in the future
(then I haven't said too much ;-).  I did set up a mailing list for
discussions related to the API (see http://com.hp.sip.listbot.com/) but
it's been very quiet.

Anders

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


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From sip-admin@lists.bell-labs.com  Tue May 30 14:46:15 2000
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From: Samandi Sami <Sami.Samandi@SRIT.siemens.fr>
To: "'Neil Deason'" <ndeason@ubiquity.net>,
        Scott Hoffpauir <scott@broadsoft.com>
Cc: sip@lists.bell-labs.com
Subject: RE: [SIP] SIP Application Server
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I have no access to all the information of the softswitch app. WG, but if I
have understood the explanation of Scott and what Jonathan answered to my
first mail, it seems that both have the same point of view. An application
server is involved in a call as any Proxy/redirect server/UAS and the
invocation is made as any routing decision via an INVITE in the very
beginning of the call. That's it.

Please correct me if I am wrong
Thank you
Sami
	-----Original Message-----
	From:	Neil Deason [SMTP:ndeason@ubiquity.net]
	Sent:	mardi 30 mai 2000 18:39
	To:	Scott Hoffpauir
	Cc:	sip@lists.bell-labs.com
	Subject:	Re: [SIP] SIP Application Server

	Thanks for providing the authoritive voice from the working group
Scott.

	Scott Hoffpauir wrote:

	> The softswitch consortium applications working group is working on
an
	> architecture supporting enhanced services in the softswitch
network
	> architecture. An application server is introduced into the
architecture to
	> host enhanced service logic and management functions. While a
softswitch
	> provides basic call control and signaling, manages resources, and
generates
	> call detail records, application servers are intended to host a
variety of
	> enhanced services. Examples of enhanced services include calling
card, 800
	> number translation, Centrex, voice mail, and find-me services. SIP
is used
	> as the interface between an application server and a softswitch.
It allows a
	> softswitch to direct calls to an application server for enhanced
services
	> processing and also allows an application server to redirect or
transfer
	> calls back through a softswitch.
	>
	> An applications server can be a proxy, redirect server or user
agent. It may
	> be stateful or stateless. It really depends on the applications
being
	> provided. I've seen applications servers which are simple redirect
servers
	> providing routing functions and more complex ones acting as
back-to-back
	> user agents, providing unified messaging. Using SIP allows a
softswitch (or
	> other end user device) to pass control to an application server,
without
	> having to worry about its architecture or functions.

	My apologies if I am confusing the terms here but am I wrong in
thinking that a
	SIP Proxy Server qualifies as a Softswitch? I have seen them
described as such
	within the Softswitch Consortium. If a SIP Proxy is a Softswitch,
isn't this
	approach in danger of reproducing the SSP/SCP model? Why would a
Proxy Server
	need to pass an INVITE on to an Application Server for the AS to
then
	proxy/redirect it? Couldn't SIP-CGI and CPL provide ways for a Proxy
to access
	any service logic required to decide whether to proxy/redirect it
directly. A
	Proxy Server augmented with service logic seems to be both a
Softswitch and an
	Application Server, which would make defining an interface based on
SIP between
	the two interesting.

	Regards,
	Neil.
	--
	Ubiquity Software Corporation, UK          http://www.ubiquity.net

	> Jonathan Rosenberg wrote:
	>
	> > Samandi Sami wrote:
	> > >
	> > > Hello,
	> > >
	> > > Is there any document (or work) out there describing the
distribution of
	> > > logic between a feature server (or application server) and the
	> underlying
	> > > network server asking for the services.
	>
	> The Softswitch consortium's Application Working Group are
interested in this
	> area
	> you might want to also look into their work ...
	>          http://www.softswitch.org/Working_Groups/application.html
	>
	> > It is true this is an implementation
	> > > issue but it is also true that this split of the roles between
a SIP
	> > > application server and a SIP proxy server is similar to the IN
	> architecture
	> > > where the SSP handles the call control and the SCP the service
control.
	> >
	> > The difference between and applications server and a proxy is
only in
	> > the complexity of the programs which guide their operation.
	> >
	> > Both speak SIP; both can proxy or redirect; there is no way to
	> > distinguish a proxy from an applications server from the
outside. This
	> > is quite different from the SSP/SCP model, which presumes
totally
	> > different interfaces and roles.
	>
	> I am not part of the Softswitch Application Working Group, and
certainly do
	> not
	> speak for them, but to me this doesn't appear to totally fit with
what I
	> understand to be their position. I know there are people on this
list close
	> to
	> this work, perhaps they would care to comment.
	>
	> The Softswitch Application Group are seeking to define the
interfaces
	> between an
	> Application Server and a Softswitch. As I understand it a SIP
Proxy Server
	> can be
	> one example of a Softswitch. They have described the Softswitch as
providing
	> basic
	> call control and signalling, managing resources, and generating
call detail
	> records. Whereas application servers host the service logic for a
variety of
	> enhanced services and media servers provide bearer oriented
functions, such
	> as
	> IVR, conferencing, and fax. They suggest SIP is used as the
interface
	> between an
	> Application Server and a Softswitch, while the Real Time Protocol
(RTP) is
	> used as
	> the interface between a media server and media gateway.
	>
	> Please note I am not seeking to promote the Softswitch view over
the one
	> described
	> by Jonathan, which I personally empathise with, just some
discussion of a
	> very
	> interesting area.
	>
	> Regards,
	> Neil
	> --
	> Ubiquity Software Corporation, UK          http://www.ubiquity.net
	>
	> > I think the SIP model is a very powerful
	> > model. It means that decisions about service invocation can
always be
	> > represented as routing decisions. If I want some service from
provider
	> > X, all that needs to happen is to have my call routed to the
server at
	> > domain X.
	> >
	> > One of the reasons the separation between call control and
service
	> > control was made in the IN, was the desire for the SCP not to
receive
	> > all signaling messages for the call. I will note that in SIP,
this
	> > function is already provided in the call signaling itself,
through the
	> > Record-Route mechanism. This means an application server can
drop out of
	> > the call after routing the initial request, as can a proxy.
	> >
	> >  Is
	> > > it then logical to think about an implementation like the
following :
	> > >
	> > > -       The network server is a call stateful proxy (holding
the state
	> of a
	> > > call between the INVITE and the BYE). This SIP proxy forward
requests to
	> the
	> > > application server. This proxy server goes stateful and proxy
requests
	> only
	> > > when needed (for instance when the end user has subscribed to
predefined
	> > > services).
	> >
	> > Why? As I argue above, this distinction of an SSP/SCP role makes
less
	> > sense in SIP.
	> >
	> > > -       The application server will behave per call basis
whether as a
	> > > redirect server (to provide routing services) or as a UAS when
media
	> > > services are needed (IVR, ...). I am not sure whether this
equipment
	> should
	> > > also proxy the request ?.
	> >
	> > If that is part of the service, sure.
	> >
	> > > -       The goal here is not to involve the application server
for the
	> whole
	> > > duration of the call but only when needed and this means
telling the
	> > > application server what's state of the call that triggered the
call of
	> the
	> > > service. For instance, A call B, but  B is busy,  the Proxy
server of B
	> > > receives a 486 and then sends an INVITE to the feature server
asking
	> what to
	> > > do. How to tell the application server that the user is busy
(the
	> equivalent
	> > > of the DP Terminal busy of the IN) ? Should this be done in a
	> proprietary
	> > > way ?
	> >
	> > I would not do it this way. I would send the initial INVITE to
the
	> > feature server, and when the 486 response comes, have it go from
there.



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From sip-admin@lists.bell-labs.com  Tue May 30 15:19:23 2000
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From: "Roy, Radhika R, ALARC" <rrroy@att.com>
To: Anders Kristensen <ak@hplb.hpl.hp.com>, Tricia Wang <TWang@NeTrue.com>
Cc: sip@lists.bell-labs.com
Subject: RE: [SIP] Domain or Zone concept
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Hi, Anders:

Does it NOT mean that we should NOT use the term like "SIP Domain, SIP
Intra-domain, or SIP Inter-Domain?"

However, it appears that the terms like administrative domains, trust
domains, inter-domain QoS, etc. as well as domains based on DNS are being
used.

Is there any way that we can use the term "domain" in a consistent way
across all references?

Best regards,
Radhika R. Roy

> -----Original Message-----
> From:	Anders Kristensen [SMTP:ak@hplb.hpl.hp.com]
> Sent:	Tuesday, May 30, 2000 2:32 PM
> To:	Tricia Wang
> Cc:	sip@lists.bell-labs.com
> Subject:	Re: [SIP] Domain or Zone concept
> 
> 
> Tricia Wang wrote:
> > 
> > Hi,
> >    does SIP have Domain or Zone concept ?
> > I would assume it does, cause I saw a Draft describing SIP interdomain
> > architecture.
> > If it does, how does it define a Domain or Zone ? by Register on a SIP
> > Server ?
> 
> I would say that the only concept of domains in vanilla SIP is really
> those of DNS. At a different level people talk about a particular
> providers network consituting a domain of its own, especially when
> discussing trust issues. Trust domains will obviously not generally map
> onto DNS domains one-to-one.
> 
> > 
> > By the way, is there a API for SIP network?  I saw a Draft Java Servlet
> API
> > but that's about it.
> 
> Right. That would be
> 
> http://www.ietf.org/internet-drafts/draft-kristensen-sip-servlet-00.txt
> 
> There will hopefully be an update of that draft sometime in the future
> (then I haven't said too much ;-).  I did set up a mailing list for
> discussions related to the API (see http://com.hp.sip.listbot.com/) but
> it's been very quiet.
> 
> Anders
> 
> -- 
> Anders Kristensen <ak@hplb.hpl.hp.com>,
> http://www-uk.hpl.hp.com/people/ak/
> Hewlett-Packard Labs, Bristol, UK
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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From sip-admin@lists.bell-labs.com  Tue May 30 15:39:34 2000
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--------------15E65F369EB87EFDDB174DCC
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Which open souce SIP code do you have?  The vovida open source SIP code  has
been ported to VXWORKS.


Also, you may find more information in vovida mailing list
sip@lists.vovida.com ;   if you have questions specific to vovida SIP stack.

Hope that helps
sunitha


Nishith Chudasama wrote:

> Hi all,
>
> I am about to start working on the open source SIP code.
>
> Can anybody tell me, which all OSes are supported by the code except LINUX?
>
> I had an overview of the code and could find some lines with "#ifdef
> VXWORKS".
> Has this code been ported to VXWORKS?
>
> Thanks & Regards,
> Nishith.
>
> ____________________________________________________________________
> Get free email and a permanent address at http://www.netaddress.com/?N=1
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--
Sunitha Kumar
http://www.vovida.com



--------------15E65F369EB87EFDDB174DCC
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Which open souce SIP&nbsp;code do you have?&nbsp; The vovida open source
SIP code&nbsp; has been ported to VXWORKS.
<br>&nbsp;
<p>Also, you may find more information in vovida mailing list&nbsp; sip@lists.vovida.com
;&nbsp;&nbsp; if you have questions specific to vovida SIP stack.
<p>Hope that helps
<br>sunitha
<br>&nbsp;
<p>Nishith Chudasama wrote:
<blockquote TYPE=CITE>Hi all,
<p>I am about to start working on the open source SIP code.
<p>Can anybody tell me, which all OSes are supported by the code except
LINUX?
<p>I had an overview of the code and could find some lines with "#ifdef
<br>VXWORKS".
<br>Has this code been ported to VXWORKS?
<p>Thanks &amp; Regards,
<br>Nishith.
<p>____________________________________________________________________
<br>Get free email and a permanent address at <a href="http://www.netaddress.com/?N=1">http://www.netaddress.com/?N=1</a>
<p>_______________________________________________
<br>SIP mailing list
<br>SIP@lists.bell-labs.com
<br><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></blockquote>

<pre>--&nbsp;
Sunitha Kumar
<A HREF="http://www.vovida.com">http://www.vovida.com</A></pre>
&nbsp;</html>

--------------15E65F369EB87EFDDB174DCC--



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Subject: Re: [SIP] Allowing the user@host form on Via header
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Comments inline

> I propose the following changes to draft-ietf-sip-2543bis-00:
> 
> Page 59, BNF of sent-by for Via header should read:
> sent-by = ([user@]host[:port])|(concealed-host)
> where user is as defined in Section 2, on page 18.
> 
> Also, in section 4.2.6, page 28, description of Request-URI, delete the
> sentence "The user name MUST be empty."
> 
> The reason for both changes is that we have a model in which a SIP Server
> is providing proxy services to a large collection of users.  It often
> happens that a client belonging to user1 will send a SIP request such as
> an INVITE to this server.  The server will receive this request and make
> various decisions according to logic installed on behalf of user1.  This
> logic frequently requires stateful interactions.  One decision could be to
> proxy the INVITE to the server for the invited party (user2).  The proxy
> server for user2 is located on the same host (and same port) as the server
> for user1.  The server for user2 may create the response to the INVITE or
> it may proxy it onwards to one or more clients for user2, or perhaps to
> other users that have been designated by user2.  This is all according to
> logic installed on behalf of user2. When the response comes back to the
> server we want the server for user2 to see the response (and perhaps
> decide to proxy the request to SIP entities) before the response is seen
> by the server for user1.  The addition of the user part in the sent-by of
> the via allows such a routing.  If it is ever used in a situation where
> there is no concept of different servers for different users it is
> ignored.
>
> Note that the branch parameter does not meet our needs since there may, in
> fact, be no branching.  Also, note that a common port on the SIP Server
> services all user agents, so the port parameter could not be used to
> differentiate servers for different users.  And, there may be more users
> serviced on this host than could be separated with a typical 16 bit port
> number.
>

You would only need to play these games if the proxy you are referring to is
transaction-stateless. If it is NOT transaction-stateless, then it is a trivial
matter to add the user information from the Request-URI to the transaction state
you are already maintaining. Frankly for the application you have in mind, it
makes more sense to use a stateful proxy than to try and change the way SIP Via:
handling works. (Though I think your use of the term "proxy" is a little different
from what I mean given your assumption that you have two proxies on the same machine
using the same port; in your case, the routing you are doing seems to be an internal
application decision and doesn't need to be explicitly handled by the SIP protocol)

 
> The suggestion for removing the sentence about the user part in the
> Request-URI of REGISTER requests is similarly motivated. There is no
> reason given in the rfc why the user part MUST be empty in a Request-URI
> for REGISTER.  In our model, a client will want to REGISTER itself with
> the server representing it within the SIP Server.  In practice, it would
> be sufficient to use the From or To field of the REGISTER to determine
> which server should be used, but in a large system, where the servers
> might be distributed among many hosts (even though they are seen as a
> single SIP server host to the outside world) it is more efficient to use
> the Request-URI for routing purposes so we want the user name to be part
> of the Request-URI.
> 

Though I don't agree with this statement, I do agree that the user name should
be allowed in the Request-URI.


> Ray Zibman
> Senior Scientist
> MetaTel, Inc.
> 45 Rumford Ave.
> Waltham, MA 02453
> (781) 392-2509
> 

--
Sean Olson <sean.olson@ericsson.com>
tel: 972-583-5472


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From sip-admin@lists.bell-labs.com  Tue May 30 16:27:48 2000
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"Roy, Radhika R, ALARC" wrote:
> 
> Hi, Anders:
> 
> Does it NOT mean that we should NOT use the term like "SIP Domain, SIP
> Intra-domain, or SIP Inter-Domain?"

Yes, FWIW I don't think those terms are well-defined and thus should be
qualified somehow.

> 
> However, it appears that the terms like administrative domains, trust
> domains, inter-domain QoS, etc. as well as domains based on DNS are being
> used.

Right, but in these cases the qualifications of the term "domain" gives
some sort of idea of what's meant, so that's more reasonable.

> 
> Is there any way that we can use the term "domain" in a consistent way
> across all references?

Just because RFC2543 doesn't talk about administrative domains doesn't
mean they are not important. I don't think there is any need for a
language police looking after how these terms are used (which is
probably not what you meant either), except to encourage people to be
clear about what they mean when they use them.

I feel a quote coming on...

  "Simplicity, clarity, singleness: These are the attributes that give
our lives power and vividness and joy as they are also the marks of
great art."
				-- Richard Holloway

just looked that up. Should be in the SIP spec, really ;-)

Anders

> 
> Best regards,
> Radhika R. Roy
> 
> > -----Original Message-----
> > From: Anders Kristensen [SMTP:ak@hplb.hpl.hp.com]
> > Sent: Tuesday, May 30, 2000 2:32 PM
> > To:   Tricia Wang
> > Cc:   sip@lists.bell-labs.com
> > Subject:      Re: [SIP] Domain or Zone concept
> >
> >
> > Tricia Wang wrote:
> > >
> > > Hi,
> > >    does SIP have Domain or Zone concept ?
> > > I would assume it does, cause I saw a Draft describing SIP interdomain
> > > architecture.
> > > If it does, how does it define a Domain or Zone ? by Register on a SIP
> > > Server ?
> >
> > I would say that the only concept of domains in vanilla SIP is really
> > those of DNS. At a different level people talk about a particular
> > providers network consituting a domain of its own, especially when
> > discussing trust issues. Trust domains will obviously not generally map
> > onto DNS domains one-to-one.
> >
> > >
> > > By the way, is there a API for SIP network?  I saw a Draft Java Servlet
> > API
> > > but that's about it.
> >
> > Right. That would be
> >
> > http://www.ietf.org/internet-drafts/draft-kristensen-sip-servlet-00.txt
> >
> > There will hopefully be an update of that draft sometime in the future
> > (then I haven't said too much ;-).  I did set up a mailing list for
> > discussions related to the API (see http://com.hp.sip.listbot.com/) but
> > it's been very quiet.
> >
> > Anders
> >

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


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Jo Hornsby wrote:
> 
> Gentlemen,
> 
> > > > Presumably, this function (TCP/UDP translation) would be provided
> > > > only if the clients couldn't talk to each other directly (e.g.
> > > > one is TCP only, and the other is UDP only). In this case, the
> > > > proxy must remain in the path to continue to provide TCP/UDP
> > > > translation for the duration of the call leg.
> > >
> > > Since then, some additional discussion seems to indicate that things
> > > get a lot easier if we require UAs to speak UDP as the lingua franca.
> > > Otherwise, we can't satisfy the goal of simply plugging in a bunch of
> > > SIP phones and setting up calls, without a proxy.
> 
> So are we saying that adding a Record-Route header is not necessary
> just because a proxy is acting as a TCP/UDP bridge?

Yes. 

> 
> > To be specific, the proposal is that UDP support in UA transition from
> > SHOULD to MUST, so that:
> >
> > 1. UAs SHOULD support TCP, MUST support UDP
> > 2. proxies MUST support TCP and UDP
> >
> > The transport parameter in the Contact and Record-Route headers then
> > becomes the preferential choice; if not supported, you can always fall
> > back to UDP.
> 
> Hmmm... isn't the implication of this that servers should ensure
> that for every TCP port listened on (apologies for the
> socket-oriented terminology), they have also bound a UDP socket
> with the same port number?  Furthermore, these ports should have
> the same semantics?  (I can imagine a proxy that exhibits different
> behaviour on different ports.)

Yes, I suppose this is the implication. This is needed even if you
assume everyone doesn't speak UDP, and then proxies which translate must
record-route. 

> 
> Is this ever going to be a problem?  (I suspect not, but it prob.
> needs to be stated, if not already.)

If you want different behavior on different ports, thats fine. All it
means is that on a particular port, the behavior for UDP and TCP would
need to be the same.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Tue May 30 17:58:50 2000
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Plong@smithmicro.com wrote:
> 
> Dean,
> 
> So is this statement incorrect? Section 1.3: "A session as defined for SDP
> can comprise one or more RTP sessions."

This is a different question. I think SDP requires a single m line,
although I think this has also been the subject of some discussion...

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Tue May 30 18:00:56 2000
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The statement is CORRECT -- a session can comprise one or more RTP
sessions. However, it is not COMPLETE -- a session can comprise one or
more network sessions of any sort established by SIP, including RTP and
other protocols.

Historically, there has been great interest in using SIP for voice
communications, hence the emphasis on SDP. But from a design
perspective, it is not limited to RTP. Foe example, I saw last year a
demo of Quake initiation using SIP. 

--
Dean

Plong@smithmicro.com wrote:
> 
> Dean,
> 
> So is this statement incorrect? Section 1.3: "A session as defined for SDP
> can comprise one or more RTP sessions."
> 
> Paul Long (an admitted hair splitter)
> Smith Micro Software, Inc.


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Neil Deason wrote:
> 
> > An applications server can be a proxy, redirect server or user agent. It may
> > be stateful or stateless. It really depends on the applications being
> > provided. I've seen applications servers which are simple redirect servers
> > providing routing functions and more complex ones acting as back-to-back
> > user agents, providing unified messaging. Using SIP allows a softswitch (or
> > other end user device) to pass control to an application server, without
> > having to worry about its architecture or functions.
> 
> My apologies if I am confusing the terms here but am I wrong in thinking that a
> SIP Proxy Server qualifies as a Softswitch? I have seen them described as such
> within the Softswitch Consortium. If a SIP Proxy is a Softswitch, isn't this
> approach in danger of reproducing the SSP/SCP model? Why would a Proxy Server
> need to pass an INVITE on to an Application Server for the AS to then
> proxy/redirect it? Couldn't SIP-CGI and CPL provide ways for a Proxy to access
> any service logic required to decide whether to proxy/redirect it directly. A
> Proxy Server augmented with service logic seems to be both a Softswitch and an
> Application Server, which would make defining an interface based on SIP between
> the two interesting.

This leads back to my original statement. The only difference between a
proxy (I won't get into the softswitch vs. proxy thing, which is purely
a definition issue) and an application server is the complexity of the
programs it executes. In the case you describe, of a proxy implementing
CGI or CPL or JAIN or parlay or whatever apps, it is then an app server.
One could even argue that the proxy function is a type of application
running on an app server. Fact is, it doesn't really matter. 

Samandi Sami later writes:
> I have no access to all the information of the softswitch app. WG, but if I
> have understood the explanation of Scott and what Jonathan answered to my
> first mail, it seems that both have the same point of view. An application
> server is involved in a call as any Proxy/redirect server/UAS and the
> invocation is made as any routing decision via an INVITE in the very
> beginning of the call. That's it.

Yes. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Jo Hornsby wrote:
> 
> Hi,
> 
> (SIP List: Apologies for the use of "Gentlemen" in my last post;
>  very poor indeed.)
> 
> > I've just submitted an updated version of the I-D on "Reliability of
> > Provisional Responses in SIP". Until it appears in the archives, you
> > can find a copy at:
> [...]
> 
> Well it's certainly easier for a proxy than the original draft (00).
> Which makes me happy. &:)
> 
> One thing I'm a little confused about: Section 5.2 (UAS Behavior)
> contains the following text:
>    "Note that a UAS can send reliable provisional responses
>     for any request, including a PRACK request. It is anticipated
>     that reliable provisional responses will be most useful for
>     INVITE requests."
> Is allowing a PRACK to be responded to with reliable provisional
> responses potentially a big old nightmare? 

Only if you implement it wrong ;)

The whole idea is that PRACK is a request, like any other. If you don't
treat it differently than any other request, in terms of its ability to
be responded to, you have no problem. 

However, if people want to nix the ability to send reliable provisional
responses to PRACK itself, I am willing to discuss. I doubt its useful.


 Or am I just being a
> little bit dull, really?  Also, shouldn't it be stated that
> reliable provisional responses can't be used with ACK?

ACK can never have responses; this draft doesn't change that. I don't
think its confusing, but I'll add some verbiage to that effect.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Tue May 30 18:23:06 2000
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Date: Tue, 30 May 2000 18:35:47 -0400
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Lars Berggren wrote:
> 
> I quote:
> "We define the calling URI as the URI found in the Contact header of the
>  request from the calling UA, if present, or the From header, if there is
>  no Contact header. The called UA copies the Record-Route entries,
>  maintaining their ordering, to the Route header field. However, all
>  components except the maddr parameter are replaced by the
>  calling URI."
> 
> As I interpret this, a proxy can not specify which port or the transport
> protocol it wants to be used in subsequent requests from the called UA
> since the only parameter of the url of the proxy that is reflected in the
> Route-list is maddr. Is this correct?

Actually, I think this should include port and transport (although as
per other threads the transport is not terribly critical, more a
preference for a first try).

> 
> What about rr-extension, is it supposed to be copied into the route field
> as a route-extension or is that supposed to be defined by the extension?

THis came up at the bakeoff. Copying it into the Route header doesn't
help, since that Route header is stripped off before getting back to the
person who inserted it. Thus, you can put something into Record-Route,
but you won't get it back in Route. This is what the State header in DCS
is for.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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kbinu@hss.hns.com wrote:
> 
> >So, if a server receives an INVITE and it responses with 400 for
> >instance. If a CANCEL is received, it will stop sending the 400 and it
> >will send the 487. It will retransmit the 487 until the ACK for the
> >INVITE arrives. Does the retransmission counter get re-started (the
> >server might have retransmitted already some times the 400 response)?
> 
> So the correct behaviour here would be to stop the 400 response
> retransmissions for the INVITE and answer the CANCEL with a 200 OK. The
> server would send a 487 if the cancelled INVITE happens to come by and not
> otherwise.

No, no, no. 

This section of the bis draft has not changed, but it will. We have
discussed this on the list extensively. If you receive a request, and
you haven't yet sent a final response, and then a CANCEL comes, respond
to the CANCEL with 200 OK, and the original request with 487. Continue
to retransmit that 487 as you normally would , even if a BYE comes.

If you've already responded to a request with a final response, and then
a CANCEL comes, respond to the CANCEL with 200 OK, and thats it.
Continue with processing the final response to that request
independently of the CANCEL.

No matter what, don't ever start sending one final response, and then
change to a different once when CANCEL comes. Very bad.

The idea here is that CANCEL is a separate transaction, which basically
says "respond to this original request with 487 if you haven't already,
otherwise don't do anything special". The CANCEL and the transaction it
cancels still complete normally. No matter what, each transaction should
execute and complete normally, independent of any other. 

This includes the case where you get an INVITE, and then a BYE. Respond
to the INVITE with a 487 (or something), and then respond to the BYE
with 200 OK. Both the INVITE and BYE transactions complete normally.

We are in the process of updating this diagram to reflect this.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Tue May 30 19:20:30 2000
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As some of you may have noticed, my office mailing address is now:

	dwillis@dynamicsoft.com

However, you may also see me writing from (or send mail to me at)

	dean.willis@softarmor.com
	dwillis@softarmor.com
	dwillis@greycouncil.com

--
Dean


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Date: Tue, 30 May 2000 20:11:45 -0400
From: Billy Biggs <vektor@div8.net>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: kbinu@hss.hns.com, Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>,
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Jonathan Rosenberg (jdrosen@dynamicsoft.com):

> The idea here is that CANCEL is a separate transaction, which basically
> says "respond to this original request with 487 if you haven't already,
> otherwise don't do anything special". The CANCEL and the transaction it
> cancels still complete normally. No matter what, each transaction should
> execute and complete normally, independent of any other. 
> 
> This includes the case where you get an INVITE, and then a BYE. Respond
> to the INVITE with a 487 (or something), and then respond to the BYE
> with 200 OK. Both the INVITE and BYE transactions complete normally.

  I do hate to beat on this more, but...

  In the case where you get an INVITE and then a BYE with a higher CSeq,
it seems really silly to have to keep around state for both of these
transactions on the call leg.  Why bother with the INVITE transaction at
all?  The far end has cancelled it anyways?  That just seems like more
unnecessary state to keep in my code.

  It seems more important to me to worry about keeping the endpoints
simple rather than the proxies, otherwise it's going to take everyone
alot longer to implement good clients.  Proxies can afford more bloat
than endpoints: let them time out or be smarter with the CSeqs.

  Any proxy that cares that much about old transaction state is likely
subject to alot of DoS attacks anyways.


  Since that argument will likely get me nowhere, please hear the message
that I would like to see simpler endpoint requirements.

-- 
Billy Biggs                         vektor@div8.net
http://www.div8.net/billy       wbiggs@uwaterloo.ca


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Billy Biggs wrote:
> 
> Jonathan Rosenberg (jdrosen@dynamicsoft.com):
> 
> > The idea here is that CANCEL is a separate transaction, which basically
> > says "respond to this original request with 487 if you haven't already,
> > otherwise don't do anything special". The CANCEL and the transaction it
> > cancels still complete normally. No matter what, each transaction should
> > execute and complete normally, independent of any other.
> >
> > This includes the case where you get an INVITE, and then a BYE. Respond
> > to the INVITE with a 487 (or something), and then respond to the BYE
> > with 200 OK. Both the INVITE and BYE transactions complete normally.
> 
>   I do hate to beat on this more, but...
> 
>   In the case where you get an INVITE and then a BYE with a higher CSeq,
> it seems really silly to have to keep around state for both of these
> transactions on the call leg.  Why bother with the INVITE transaction at
> all?  The far end has cancelled it anyways?  That just seems like more
> unnecessary state to keep in my code.

Its good for UAs, and really good for proxies. For the UA, it allows for
independence of transactions, and separation of transaction processing
from call state. I would argue that requiring the INVITE transaction to
cease

For proxies, which don't see call state in many cases, it allows for
independent operation of each transaction. Without that, proxies will be
left with state hanging around (i.e., INVITE transaction state) when the
BYE comes, particularly when the proxy doesn't even see the BYE. This
leads to wasted resources in proxies, performance bottlenecks, and more
complex code to clean up.


> 
>   It seems more important to me to worry about keeping the endpoints
> simple rather than the proxies, otherwise it's going to take everyone
> alot longer to implement good clients.  Proxies can afford more bloat
> than endpoints: let them time out or be smarter with the CSeqs.

I disagree completely. What you are suggesting is at odds with the
entire IP model of scalability (and with the model supported by SIP),
which says to push complexity to the edge, where it is easier to manage.
In any case, this detail of completing INVITE transactions hardly
qualifies as complex. As I argue above, I think it makes things simpler.


> 
>   Any proxy that cares that much about old transaction state is likely
> subject to alot of DoS attacks anyways.

You want your protocols to clean up state under normal operation.


-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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On the contrary, the fact that I let all transactions complete their normal
way has imho, made our client development
much easier. This has meant that our transaction request-response code
remains usually intact, and hence,  we do not
really have to maintain cross-related state - only one transaction state
and still be able to support BYE and more importantly CANCEL.

Further, there is a bigger issue for not having the transaction complete as
you mention.
Lets suppose client A sends an INVITE to client B through proxy P.
Now again, lets assume that B returned A a contact address for further
requests - which happens to be the direct address of B.
Then that would mean that A would send its future request directly to B
without P (Assuming a simple proxy that does not do
record-route).

What would then happen is that when A sent a BYE directly to B , the proxy
would have no idea that the pending transaction
of INVITE is no longer valid - and would not have any option but to wait
for a regular timeout to happen before it could flush
the old INVITE out. This is certainly not a good situation.

The beauty, imho, of letting *all* transactions complete is that no
inbetween node is left hanging in between a transaction
state unnecessarily.

The merits of transaction completeness, I feel, over-weigh message
optimisation.

Regs
Arjun
--
Arjun Roychowdhury @ Hughes Software Systems


Billy Biggs wrote:


 I do hate to beat on this more, but...

  In the case where you get an INVITE and then a BYE with a higher CSeq,
it seems really silly to have to keep around state for both of these
transactions on the call leg.  Why bother with the INVITE transaction at
all?  The far end has cancelled it anyways?  That just seems like more
unnecessary state to keep in my code.

  It seems more important to me to worry about keeping the endpoints
simple rather than the proxies, otherwise it's going to take everyone
alot longer to implement good clients.  Proxies can afford more bloat
than endpoints: let them time out or be smarter with the CSeqs.




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Subject: Re: [SIP] Basic questions
Date: Wed, 31 May 2000 01:49:22 -0400
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Jonathan,

SDP does not mandate a single m= line.  Am I missing something?  It's
perfectly valid to do this in SDP I believe:

   m=audio 2500 RTP/AVP 0
   m=video 2502 RTP/AVP 31

The ABNF grammar allows it and there are several examples in RFC 2327.

Paul

----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: <Plong@smithmicro.com>
Cc: <dwillis@dynamicsoft.com>; <sip@lists.bell-labs.com>
Sent: Tuesday, May 30, 2000 6:11 PM
Subject: Re: [SIP] Basic questions


>
>
> Plong@smithmicro.com wrote:
> >
> > Dean,
> >
> > So is this statement incorrect? Section 1.3: "A session as defined for
SDP
> > can comprise one or more RTP sessions."
>
> This is a different question. I think SDP requires a single m line,
> although I think this has also been the subject of some discussion...
>
> -Jonathan R.
>
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
>
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>



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Jonathan D. Rosenberg wrote 

	If you've already responded to a request with a final response, and then
	a CANCEL comes, respond to the CANCEL with 200 OK, and thats it.
	Continue with processing the final response to that request
	independently of the CANCEL.

Dont we have to send a 481 response to the CANCEL as the earlier
transaction has been completed as a final response has already  been sent. 

In the same case if the server were a proxy, then does it not forward the
CANCEL downstream too?

 Srinath A
 Silicon Automation Systems       Phone: 5276100, 5276108 extn 4220
 #1309, 10th Main, HAL III Stage   
 Bangalore 560 008, India                    



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Sorry; I meant at least a single m line (i.e., zero m lines not
allowed).

-Jonathan R.

"Paul E. Jones" wrote:
> 
> Jonathan,
> 
> SDP does not mandate a single m= line.  Am I missing something?  It's
> perfectly valid to do this in SDP I believe:
> 
>    m=audio 2500 RTP/AVP 0
>    m=video 2502 RTP/AVP 31
> 
> The ABNF grammar allows it and there are several examples in RFC 2327.
> 
> Paul
> 
> ----- Original Message -----
> From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
> To: <Plong@smithmicro.com>
> Cc: <dwillis@dynamicsoft.com>; <sip@lists.bell-labs.com>
> Sent: Tuesday, May 30, 2000 6:11 PM
> Subject: Re: [SIP] Basic questions
> 
> >
> >
> > Plong@smithmicro.com wrote:
> > >
> > > Dean,
> > >
> > > So is this statement incorrect? Section 1.3: "A session as defined for
> SDP
> > > can comprise one or more RTP sessions."
> >
> > This is a different question. I think SDP requires a single m line,
> > although I think this has also been the subject of some discussion...
> >
> > -Jonathan R.
> >
> > --
> > Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> > Chief Scientist                             First Floor
> > dynamicsoft                                 East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> > http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> > http://www.dynamicsoft.com
> >
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> >

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Jonathan,

If you need to have at least a single m line, why is it written page 8 of the
RFC 2327 (SDP) that
you can have ZERO or more media descriptions ?

-Alain

Jonathan Rosenberg wrote:

> Sorry; I meant at least a single m line (i.e., zero m lines not
> allowed).
>
> -Jonathan R.
>



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Subject: Re: [SIP] wg last call on reliable provisional responses
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Jonathan Rosenberg wrote:

> Jo Hornsby wrote:
> >
> > Is allowing a PRACK to be responded to with reliable provisional
> > responses potentially a big old nightmare?
>
> Only if you implement it wrong ;)

I think what is being hinted at albeit subtlety is, is the transaction ever
going to end? (The Halting Problem.) Although I can't think of a case off
hand where you'd want to provisionally respond to a PRACK, specifically
allowing it is just asking for trouble between legitimate implementations.

>
> The whole idea is that PRACK is a request, like any other. If you don't
> treat it differently than any other request, in terms of its ability to
> be responded to, you have no problem.
>

Except it may spawn another PRACK that spawns another PRACK ad infinitum.

>
> However, if people want to nix the ability to send reliable provisional
> responses to PRACK itself, I am willing to discuss. I doubt its useful.
>

I'd vote for cutting it simply for the reason mentioned above.

James Undery



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Jonathan Rosenberg wrote:

> Sorry; I meant at least a single m line (i.e., zero m lines not
> allowed).
>

4th paragraph of section 6 of RFC 2327 starts "An announcement consists of a
session-level section followed by zero or more media-level sections". While this
may be useful in a generic case the lack of media means the it is of no use to
SIP. I would suggest adding a line along the lines of, SDP used in SIP must
contain one or more media-level descriptions, to the bis draft. (Obviously
reworded to avoid my awkward prose ;^) )


James Undery



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From sip-admin@lists.bell-labs.com  Wed May 31 04:19:41 2000
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Date: Wed, 31 May 2000 09:15:40 +0100
From: Neil Deason <ndeason@ubiquity.net>
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Jonathan Rosenberg wrote:

> Neil Deason wrote:
> >
> > > An applications server can be a proxy, redirect server or user agent. It may
> > > be stateful or stateless. It really depends on the applications being
> > > provided. I've seen applications servers which are simple redirect servers
> > > providing routing functions and more complex ones acting as back-to-back
> > > user agents, providing unified messaging. Using SIP allows a softswitch (or
> > > other end user device) to pass control to an application server, without
> > > having to worry about its architecture or functions.
> >
> > My apologies if I am confusing the terms here but am I wrong in thinking that a
> > SIP Proxy Server qualifies as a Softswitch? I have seen them described as such
> > within the Softswitch Consortium. If a SIP Proxy is a Softswitch, isn't this
> > approach in danger of reproducing the SSP/SCP model? Why would a Proxy Server
> > need to pass an INVITE on to an Application Server for the AS to then
> > proxy/redirect it? Couldn't SIP-CGI and CPL provide ways for a Proxy to access
> > any service logic required to decide whether to proxy/redirect it directly. A
> > Proxy Server augmented with service logic seems to be both a Softswitch and an
> > Application Server, which would make defining an interface based on SIP between
> > the two interesting.
>
> This leads back to my original statement. The only difference between a
> proxy (I won't get into the softswitch vs. proxy thing, which is purely
> a definition issue) and an application server is the complexity of the
> programs it executes. In the case you describe, of a proxy implementing
> CGI or CPL or JAIN or parlay or whatever apps, it is then an app server.
> One could even argue that the proxy function is a type of application
> running on an app server. Fact is, it doesn't really matter.
>
> Samandi Sami later writes:
> > I have no access to all the information of the softswitch app. WG, but if I
> > have understood the explanation of Scott and what Jonathan answered to my
> > first mail, it seems that both have the same point of view. An application
> > server is involved in a call as any Proxy/redirect server/UAS and the
> > invocation is made as any routing decision via an INVITE in the very
> > beginning of the call. That's it.
>
> Yes.

Jonathan I agree that a service logic enhanced Proxy Server is an application server.

However, I am not yet convinced this sits in complete harmony with the ISC App WG's
intention to separate the call signalling (Proxy/Softswitch) and service control (App
Server) and use standard RFC2543 SIP as the interface between them. In the case of
the service logic enhanced proxy isn't the interface between vanilla proxy and the
service control defined by CPL/SIP-CGI/Servlet API etc not RFC2543 SIP.

Regards,
Neil.
--
Ubiquity Software Corporation, UK           http://www.ubiquity.net



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From sip-admin@lists.bell-labs.com  Wed May 31 05:20:01 2000
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> Dont we have to send a 481 response to the CANCEL as the earlier
> transaction has been completed as a final response has already  
> been sent. 

No, it should return 200, since it still has knowledge of the
transaction.  There was a thread around these issues a few weeks
back.  Look for the Title "[SIP] CANCEL related questions".

> In the same case if the server were a proxy, then does it not forward
> the CANCEL downstream too?

Yes, the proxy should forward the CANCEL downstream regardless of
whether it knows anything about the transaction or not.  Again,
see the thread mentioned above.


 - Jo.



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From sip-admin@lists.bell-labs.com  Wed May 31 05:22:27 2000
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Subject: [SIP] My master thesis on SIP
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Hi SIP-people,

Just wanted to tell you that my thesis has been published, available
at http://epubl.luth.se/1402-1617/2000/137 (scroll down for pdf-file)

Not much really of interest there for someone familiar with the basics
of SIP or more, however.

/Petter


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--NextPart

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

	Title		: Reliability of Provisional Responses in SIP
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-sip-100rel-01.txt
	Pages		: 17
	Date		: 30-May-00
	
This document specifies an extension to the Session Initiation
Protocol (SIP) providing reliable provisional response messages. This
extension uses the option tag org.ietf.sip.100rel.

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Section 3, Paragraph 1: "Note that a UAS MUST NOT send a response reliably
unless there was a Supported header in the request indicating support for
this extension."
 
This should be "... unless there was a Supported or Require header in the
request indicating support for this extension."
 
-----
 
Section 3, Paragraph 3: "The UAC can determine that the response is to be
transmitted reliably by the presence of the RSeq header."
 
I believe this should read "... by the presence of a Require header
containing the option tag 100rel."  The use of the Require header in the
response is mandated (correctly, in my opinion) by Section 5.2, Paragraph 6
(UAS behavior).  This also follows draft-ietf-sip-serverfeatures-02.txt
which "allows for the Require header to appear in responses. It is used to
indicate what options must be understood by the UAC in order to process the
response."
In general, I think it's best to explicitly state what extensions/features
are being used, rather than having the user agent deduce them from the
presence/absence of headers.  This also will allow the Rseq header to be
reused in other extensions where it might make sense, rather than having to
create another extension-specific header.
 
-----
 
Section 5.1, Paragraph 2: "The rest of this discussion assumes this header
has been inserted into a request."
 
This should read "... one of these headers has been inserted ...".  The
previous paragraph was improved to allow for the UAC to insert either the
Supported header or the Require header, and this sentence needs to follow
suit.
 
-----
 
Section 5.1, Paragraph 11: "It is not necessary to include the Supported
header listing the option tag 100rel in the PRACK request."
 
I disagree.  Given that the UAS may try to send reliable provisional
responses to a PRACK (it's mentioned, and seems reasonable for completeness,
though I don't see an obvious use of this), I think the UAC ought to say
whether it supports or requires the 100rel feature.  Yes, the UAS can deduce
that the feature is supported by the presence of the PRACK method, but (1)
the UAS cannot tell from the existence of the PRACK method whether the UAC
*requires* or merely *supports* the 100rel feature, and (2) the PRACK method
could never be used in support of any other feature.  (2) might not be much
of a problem, but I think either a Require or Supported header is needed to
address (1).  Mandating the header would also make PRACK more consistent
with the other methods, all of which mandate either the Require or Supported
header.
 
-----
 
Section 5.2, Paragraph 1: "The UAS may send any provisional response
reliably, so long as the initial request contained a Supported header
indicating that this feature is understood."
 
This should read "... the initial request contained either a Supported or
Require header indicating that this feature is understood."
 
-----
 
Lastly (and maybe I've missed my chance on this one, since the "open issue"
section went away), did we ever come up with an application where the
cumulative response mechanism was needed?  I thought we had come to
consensus to drop it unless there was a demonstrated need (which I haven't
seen, though I may have just missed it).  
 
The UAC must PRACK a provisional response, which seems to disallow
accumulating responses in the hopes of sending one PRACK for the group.
Only if a provisional response arrives which is known to be out-of-sequence,
and only if it is then cached by the UAC, and only if the previous missing
response then arrives, does it seem that the cumulative response mechanism
could come into play.  The examples section uses "two provisional responses
in rapid succession" to address this, but I'm not sure that's sufficient.
When the first response arrives, I read section 5.1 (UAC behavior) to
require that the UAC construct and send a PRACK; I'm not sure waiting around
to see if another response arrives before doing so is reasonable -- it would
delay the acknowledgement of the first response in the usually vain hope
that another response would quickly arrive.  Maybe if the two responses were
sent together in a single UDP packet, the cumulative response would be
justified?  
 
Also, the last paragraph of section 3 and the penultimate paragraph of
section 5.2 both recommend that a UAS wait for acknowledgement of a
provisional response before sending the next one.  Any UAS following this
recommended procedure would certainly never need the cumulative response
mechanism.  
 
I'd take out all the verbiage dealing with the cumulative responses, in the
interests of simplicity and clarity.  There'd no longer need to be any
difference in handling between the first reliable response (which cannot be
part of a cumulative group because of the need to agree on the first Rseq)
and any further reliable responses.  Maybe the space freed up in the
examples session could be used to show an example with the Require: 100rel
header rather than the Supported: 100rel header.
 
Apologies if I missed some resolution on this cumulative response issue.
 
-----

Tim Schroeder 
tschroeder@tri.sbc.com 
SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759 

 


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From: Gethin Liddell <gethin@ubiquity.net>
Organization: Ubiquity Software Corp.
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Hi,

I think the BNF in the new bis draft concerning SipUrl and headers is wrong:

p.18 Section 2 Figure 3 says

hname = hnv-unreserved | unreserved | escaped
hvalue = hnv-unreserved | unreserved | escaped

should this not be:

hname = 1*(hnv-unreserved | unreserved | escaped)
hvalue = *(hnv-unreserved | unreserved | escaped)

small point, but just making sure

cheers

-- 
Gethin Liddell
Ubiquity Software Corporation

http://www.ubiquity.net
mailto:gethin@ubiquity.net


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From sip-admin@lists.bell-labs.com  Wed May 31 13:46:03 2000
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Subject: Re: [SIP] SipUrl header BNF incorrect in bis
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--------------A1EC5295974B319891F60598
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Further to Gethin's point, an earlier discussion on the unreserved characters
for paramaters and headers doesn't seem to have been accuarately captured in the
May bis. I think the two key requirements for these character sets are 1) they
should be subsets of the URL reserved character set (ref RFC 2396/2732), and 2)
they should avoid any characters which would cause ambiguities in the SIP
grammar.

1. It was discussed that "," should not be included in param-unreserved (and by
extension with hnv-unreserved) "because URLs are allowed in certain headers and
allowing commas in SIP URLs will make parsing those headers sometime
impossible.". I thought this point was accepted, but maybe I missed something.

2. RFC 2732 adds "[" and "]" to the URL reserved set to support IPv6 address
syntax. As a consequence, I think these should also be added to param-unreserved
and hnv-unreserved, since I don't think they cause any ambiguity issues.


Rick Workman
Nortel Networks


Gethin Liddell wrote:

> Hi,
>
> I think the BNF in the new bis draft concerning SipUrl and headers is wrong:
>
> p.18 Section 2 Figure 3 says
>
> hname = hnv-unreserved | unreserved | escaped
> hvalue = hnv-unreserved | unreserved | escaped
>
> should this not be:
>
> hname = 1*(hnv-unreserved | unreserved | escaped)
> hvalue = *(hnv-unreserved | unreserved | escaped)
>
> small point, but just making sure
>
> cheers
>
> --
> Gethin Liddell
> Ubiquity Software Corporation
>
> http://www.ubiquity.net
> mailto:gethin@ubiquity.net
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--------------A1EC5295974B319891F60598
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<tt>Further to Gethin's point, an earlier discussion on the unreserved
characters for paramaters and headers doesn't seem to have been accuarately
captured in the May bis. I think the two key requirements for these character
sets are 1) they should be subsets of the URL reserved character set (ref
RFC 2396/2732), and 2) they should avoid any characters which would cause
ambiguities in the SIP grammar.</tt><tt></tt>
<p><tt>1. It was discussed that "," should not be included in param-unreserved
(and by extension with hnv-unreserved) "because URLs are allowed in certain
headers and allowing commas in SIP URLs will make parsing those headers
sometime impossible.". I thought this point was accepted, but maybe I missed
something.</tt><tt></tt>
<p><tt>2. RFC 2732 adds "[" and "]" to the URL reserved set to support
IPv6 address syntax. As a consequence, I think these should also be added
to param-unreserved and hnv-unreserved, since I don't think they cause
any ambiguity issues.</tt>
<br><tt></tt>&nbsp;<tt></tt>
<p><tt>Rick Workman</tt>
<br><tt>Nortel Networks</tt>
<br><tt></tt>&nbsp;<tt></tt>
<p><tt>Gethin Liddell wrote:</tt>
<blockquote TYPE=CITE><tt>Hi,</tt><tt></tt>
<p><tt>I think the BNF in the new bis draft concerning SipUrl and headers
is wrong:</tt><tt></tt>
<p><tt>p.18 Section 2 Figure 3 says</tt><tt></tt>
<p><tt>hname = hnv-unreserved | unreserved | escaped</tt>
<br><tt>hvalue = hnv-unreserved | unreserved | escaped</tt><tt></tt>
<p><tt>should this not be:</tt><tt></tt>
<p><tt>hname = 1*(hnv-unreserved | unreserved | escaped)</tt>
<br><tt>hvalue = *(hnv-unreserved | unreserved | escaped)</tt><tt></tt>
<p><tt>small point, but just making sure</tt><tt></tt>
<p><tt>cheers</tt><tt></tt>
<p><tt>--</tt>
<br><tt>Gethin Liddell</tt>
<br><tt>Ubiquity Software Corporation</tt><tt></tt>
<p><tt><a href="http://www.ubiquity.net">http://www.ubiquity.net</a></tt>
<br><tt><a href="mailto:gethin@ubiquity.net">mailto:gethin@ubiquity.net</a></tt><tt></tt>
<p><tt>_______________________________________________</tt>
<br><tt>SIP mailing list</tt>
<br><tt>SIP@lists.bell-labs.com</tt>
<br><tt><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></tt></blockquote>
<tt></tt></html>

--------------A1EC5295974B319891F60598--



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From sip-admin@lists.bell-labs.com  Wed May 31 14:55:51 2000
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From: Linden deCarmo <ldeCarmo@netspeak.com>
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Subject: [SIP] Suspend a session??
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Right now, SIP smoothly handles one party putting another party on hold via
a re-INVITE with modified SDP.  However, we have a scenario where the
session needs to be suspended (not just placed on hold), and I'm not sure
how this can be done without using proprietary headers.

Here are ther details.  If party A calls party B , party B should be able to
hang up without immediately terminating the session.  If party B picks up
the phone within some prescribed time window, the session continues.
However, if they don't pick it up within the time window, the session
terminates (via a BYE).

We're looking for a mechanism for party B to communicate with either party A
or a proxy server, to inform the distant party that the session has been
suspended, and not just placed on hold.  The primary motivator for this
notification is billing.  Without notification of suspension, its impossible
to determine when to safely stop billing.

Thanks.

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



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From sip-admin@lists.bell-labs.com  Wed May 31 16:30:11 2000
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Subject: FW: [SIP] wg last call on reliable provisional responses
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James Undery writes:
I think what is being hinted at albeit subtlety is, is the transaction ever
going to end? (The Halting Problem.) Although I can't think of a case off
hand where you'd want to provisionally respond to a PRACK, specifically
allowing it is just asking for trouble between legitimate implementations.
[...] Except it may spawn another PRACK that spawns another PRACK ad
infinitum.

I was going to write in support of reliable provisional responses for PRACK,
but I think this sways me over to the opposition.  One could get one's self
into a fair amount of trouble receiving a PRACK and sending a reliable
provisional response that results in receiving another PRACK, etc.

This would also address my previous comment about mandating Require: 100rel
or Supported: 100rel in a PRACK message, rather than relying on the presence
of PRACK to imply support.  The UAC could of course include the Require or
Supported header if it chose, but it'd be moot because the UAS would not be
allowed to send a reliable provisional response.  The UAS would therefore
not be allowed to put the Require: 100rel in a provisional response to a
PRACK.

Tim Schroeder
tschroeder@tri.sbc.com
SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759


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"Dave Oran" <oran@cisco.com> on 05/30/2000 01:30:31 PM

To:   "charles Agboh" <agboh@helios.iihe.ac.be>, "Dean Willis"
      <dwillis@dynamicsoft.com>, sip@lists.bell-labs.com
cc:   "Jonathan Rosenberg" <jdrosen@dynamicsoft.com> (bcc: David K
      Brown/MAIN/MC1)

Subject:  RE: [SIP] Basic questions




> > -----Original Message-----
> > From: sip-admin@lists.bell-labs.com
> > [mailto:sip-admin@lists.bell-labs.com]On Behalf Of charles Agboh
> > Sent: Tuesday, May 30, 2000 9:06 AM
> > To: Dean Willis; sip@lists.bell-labs.com
> > Cc: Jonathan Rosenberg
> > Subject: Re: [SIP] Basic questions
> >
> >
> >
> > Can somebody please answer the following question? "
> >
> >     How can you have a "session" of any kind with only one participant?
> >
> > charles
>
> What is the sound of one hand clapping?
>
> SIP can set up Zen sessions. It's completely general.

It may be general enough to allow these sessions but are they useful (some
multi-cast situations have been identified but those looked like transitional
situations).  I'm looking at this from the standpoint of possible services, not
strictly protocol rules.  Would single particiapant sessions possibly be of use
in the following types of service conditions: transition (adding, removing, or
locating participants), residue of failed or disrupted sessions (is this
possible for even a few moments?), pending sessions, meet-me or third party
controlled type conference sessions (which might even reasonably conceptually
have NO participants at some stage).  These possible single participant
situations are all transitions in a larger service, does anyone anticipate a
single participant only service?

Dave Brown
Marconi Comms








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Subject: Re: [SIP] Suspend a session??
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Have you considered building an auto-hold on switch-hook with a timer
that initaies a BYE into your UA? My first thought is that the suggested
behavior is a UA problem, not a proxy problem. Unless it is a flaw in
the billing model . . .

B calls A.
B places receiver on-hook.
B sends reINVITE-to-HOLD to A.
B starts timer.
Timer expires.
B sends BYE to A.

The only issue I see here is that B might vanish during the hold-period
and fail to send the BYE. This is a risk with call-hold in general, and
I believe there has been a proposal to use a configurable hold-timer in
end devices to recover from it.

--
Dean

Linden deCarmo wrote:
> 
> Right now, SIP smoothly handles one party putting another party on hold via
> a re-INVITE with modified SDP.  However, we have a scenario where the
> session needs to be suspended (not just placed on hold), and I'm not sure
> how this can be done without using proprietary headers.
> 
> Here are ther details.  If party A calls party B , party B should be able to
> hang up without immediately terminating the session.  If party B picks up
> the phone within some prescribed time window, the session continues.
> However, if they don't pick it up within the time window, the session
> terminates (via a BYE).
> 
> We're looking for a mechanism for party B to communicate with either party A
> or a proxy server, to inform the distant party that the session has been
> suspended, and not just placed on hold.  The primary motivator for this
> notification is billing.  Without notification of suspension, its impossible
> to determine when to safely stop billing.
> 
> Thanks.
> 
> Linden deCarmo
> Netspeak Corporation
> 902 Clint Moore Road
> Suite 104
> Boca Raton, FL 33487
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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Date: Wed, 31 May 2000 20:46:07 -0500
From: Henry Sinnreich <Henry.Sinnreich@WCom.com>
Subject: RE: [SIP] Suspend a session??
In-reply-to: <200006011006.FAA08972@kevlar.softarmor.com>
To: Dean Willis <dwillis@dynamicsoft.com>,
        Linden deCarmo <ldeCarmo@netspeak.com>
Cc: sip@lists.bell-labs.com
Message-id: <NEBBLDFFKGAJDPBENMDNAEJHCFAA.Henry.Sinnreich@WCom.com>
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If it was my decision, I would bill only for the QoS minutes for
this call.

Henry

>-----Original Message-----
>From: sip-admin@lists.bell-labs.com
>[mailto:sip-admin@lists.bell-labs.com]On Behalf Of Dean Willis
>Sent: Wednesday, May 31, 2000 6:07 PM
>To: Linden deCarmo
>Cc: sip@lists.bell-labs.com
>Subject: Re: [SIP] Suspend a session??
>
>
>
>Have you considered building an auto-hold on
>switch-hook with a timer
>that initaies a BYE into your UA? My first thought is
>that the suggested
>behavior is a UA problem, not a proxy problem. Unless
>it is a flaw in
>the billing model . . .
>
>B calls A.
>B places receiver on-hook.
>B sends reINVITE-to-HOLD to A.
>B starts timer.
>Timer expires.
>B sends BYE to A.
>
>The only issue I see here is that B might vanish
>during the hold-period
>and fail to send the BYE. This is a risk with
>call-hold in general, and
>I believe there has been a proposal to use a
>configurable hold-timer in
>end devices to recover from it.
>
>--
>Dean
>
>Linden deCarmo wrote:
>>
>> Right now, SIP smoothly handles one party putting
>another party on hold via
>> a re-INVITE with modified SDP.  However, we have a
>scenario where the
>> session needs to be suspended (not just placed on
>hold), and I'm not sure
>> how this can be done without using proprietary headers.
>>
>> Here are ther details.  If party A calls party B ,
>party B should be able to
>> hang up without immediately terminating the session.
> If party B picks up
>> the phone within some prescribed time window, the
>session continues.
>> However, if they don't pick it up within the time
>window, the session
>> terminates (via a BYE).
>>
>> We're looking for a mechanism for party B to
>communicate with either party A
>> or a proxy server, to inform the distant party that
>the session has been
>> suspended, and not just placed on hold.  The primary
>motivator for this
>> notification is billing.  Without notification of
>suspension, its impossible
>> to determine when to safely stop billing.
>>
>> Thanks.
>>
>> Linden deCarmo
>> Netspeak Corporation
>> 902 Clint Moore Road
>> Suite 104
>> Boca Raton, FL 33487
>>
>> _______________________________________________
>> SIP mailing list
>> SIP@lists.bell-labs.com
>> http://lists.bell-labs.com/mailman/listinfo/sip
>
>
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