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Subject: [sipcore] AD Review: draft-ietf-sipcore-sip-websocket-08
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--e89a8fb1ef1634b62804d950004c
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Dear SIPCORE,

I reviewed the SIP over WebSockets draft, and think it's in good shape.  I
have requested an IETF LC.

Thanks,
--Richard

--e89a8fb1ef1634b62804d950004c
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Dear SIPCORE,<div><br></div><div>I reviewed the SIP over WebSockets draft, =
and think it&#39;s in good shape. =A0I have requested an IETF LC.</div><div=
><br></div><div>Thanks,</div><div>--Richard</div>

--e89a8fb1ef1634b62804d950004c--

From iesg-secretary@ietf.org  Mon Apr  1 10:54:10 2013
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Subject: [sipcore] Last Call: <draft-ietf-sipcore-sip-websocket-08.txt> (The WebSocket	Protocol as a Transport for the Session Initiation Protocol	(SIP)) to Proposed Standard
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The IESG has received a request from the Session Initiation Protocol Core
WG (sipcore) to consider the following document:
- 'The WebSocket Protocol as a Transport for the Session Initiation
   Protocol (SIP)'
  <draft-ietf-sipcore-sip-websocket-08.txt> as Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2013-04-15. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

Abstract


   The WebSocket protocol enables two-way realtime communication between
   clients and servers in web-based applications.  This document
   specifies a WebSocket sub-protocol as a reliable transport mechanism
   between SIP (Session Initiation Protocol) entities to enable usage of
   SIP in web-oriented deployments.  This document normatively updates
   RFC 3261.




The file can be obtained via
http://datatracker.ietf.org/doc/draft-ietf-sipcore-sip-websocket/

IESG discussion can be tracked via
http://datatracker.ietf.org/doc/draft-ietf-sipcore-sip-websocket/ballot/


No IPR declarations have been submitted directly on this I-D.



From iesg-secretary@ietf.org  Mon Apr  1 10:54:11 2013
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The IESG has received a request from the Session Initiation Protocol Core
WG (sipcore) to consider the following document:
- 'The WebSocket Protocol as a Transport for the Session Initiation
   Protocol (SIP)'
  <draft-ietf-sipcore-sip-websocket-08.txt> as Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2013-04-15. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

Abstract


   The WebSocket protocol enables two-way realtime communication between
   clients and servers in web-based applications.  This document
   specifies a WebSocket sub-protocol as a reliable transport mechanism
   between SIP (Session Initiation Protocol) entities to enable usage of
   SIP in web-oriented deployments.  This document normatively updates
   RFC 3261.




The file can be obtained via
http://datatracker.ietf.org/doc/draft-ietf-sipcore-sip-websocket/

IESG discussion can be tracked via
http://datatracker.ietf.org/doc/draft-ietf-sipcore-sip-websocket/ballot/


No IPR declarations have been submitted directly on this I-D.



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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 2 Apr 2013 11:40:26 +0200
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-sip-websocket-08
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2013/4/1 Richard Barnes <rlb@ipv.sx>:
> Dear SIPCORE,
>
> I reviewed the SIP over WebSockets draft, and think it's in good shape.  =
I
> have requested an IETF LC.


Thanks a lot Richard.

Best regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: [sipcore] Fwd: SecDir review of draft-ietf-sipcore-sip-websocket-08
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A representative from the security directorate has weighed in on his 
opinion of the SIP over WebSockets document, and believes we may have 
additional work to do. I plan to spend some time wading through the 
specific comments and determining which ones indicate a need to change 
the document versus that that merely require an explanation for Yaron's 
benefit, but it will be a while before I have time to do so.

If anyone else is in a position to perform that analysis and propose 
explanations and/or proposed document edits, please feel free to hop in 
and do so.

/a


-------- Original Message --------
Subject: 	SecDir review of draft-ietf-sipcore-sip-websocket-08
Resent-Date: 	Wed, 10 Apr 2013 16:19:04 -0500 (CDT)
Resent-From: 	draft-alias-bounces@tools.ietf.org
Resent-To: 	adam@nostrum.com, ibc@aliax.net, jmillan@aliax.net, 
pkyzivat@alum.mit.edu, rlb@ipv.sx, vpascual@acmepacket.com
Date: 	Thu, 11 Apr 2013 00:18:46 +0300
From: 	Yaron Sheffer <yaronf.ietf@gmail.com>
To: 	IETF Security Directorate <secdir@ietf.org>, The IESG 
<iesg@ietf.org>, draft-ietf-sipcore-sip-websocket.all@tools.ietf.org



I have reviewed this document as part of the security directorate's
ongoing effort to review all IETF documents being processed by the IESG.
These comments were written primarily for the benefit of the security
area directors.  Document editors and WG chairs should treat these
comments just like any other last call comments.

This document defines how SIP can be layered on the WebSocket protocol.
This is essentially a profile of SIP.

Summary

In my opinion, from a security perspective this document is not yet
ready for publication. I expect this document to be widely implemented,
and I would not want it implemented until it offers a solid security
model, even if it is optional.

Details

RFC 3261 has 20 pages of security considerations, and defines several
security mechanisms. The current document "only" defines a transport for
SIP, but such transport needs to preserve the security of any SIP
mechanisms being used. In particular, it should be clear how endpoint
identities are determined at each layer and how identities at different
layers relate to one another. In other words, what's known as "channel
binding".

The only security-related section (other than the Security
Considerations) is Authentication (Sec. 7), and this section is marked
non-normative. So the reader is left to wonder: how is the SIP client
authenticated? How does this authentication relate to the
WebSocket-level client authentication? And similarly for the server.

Sec. 9.1 mentions (again, non-normatively) that SIP-level certificate
checks are omitted, and replaced by transport-level checks only. I
believe this significantly reduces the security properties of SIP.
Moreover, it begs for a policy tying WebSocket certificates to
identities of the endpoints of the SIP protocol.

Nit/bug: Sec. 5.2.1 ends with a colon in the HTML version *only*, and is
missing a paragraph.

Thanks,
	Yaron




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<html>
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    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
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    A representative from the security directorate has weighed in on his
    opinion of the SIP over WebSockets document, and believes we may
    have additional work to do. I plan to spend some time wading through
    the specific comments and determining which ones indicate a need to
    change the document versus that that merely require an explanation
    for Yaron's benefit, but it will be a while before I have time to do
    so.<br>
    <br>
    If anyone else is in a position to perform that analysis and propose
    explanations and/or proposed document edits, please feel free to hop
    in and do so.<br>
    <br>
    /a<br>
    <div class="moz-forward-container"><br>
      <br>
      -------- Original Message --------
      <table class="moz-email-headers-table" border="0" cellpadding="0"
        cellspacing="0">
        <tbody>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:
            </th>
            <td>SecDir review of draft-ietf-sipcore-sip-websocket-08</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Resent-Date:
            </th>
            <td>Wed, 10 Apr 2013 16:19:04 -0500 (CDT)</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Resent-From:
            </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:draft-alias-bounces@tools.ietf.org">draft-alias-bounces@tools.ietf.org</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Resent-To:
            </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:adam@nostrum.com">adam@nostrum.com</a>, <a class="moz-txt-link-abbreviated" href="mailto:ibc@aliax.net">ibc@aliax.net</a>, <a class="moz-txt-link-abbreviated" href="mailto:jmillan@aliax.net">jmillan@aliax.net</a>,
              <a class="moz-txt-link-abbreviated" href="mailto:pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</a>, <a class="moz-txt-link-abbreviated" href="mailto:rlb@ipv.sx">rlb@ipv.sx</a>, <a class="moz-txt-link-abbreviated" href="mailto:vpascual@acmepacket.com">vpascual@acmepacket.com</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
            <td>Thu, 11 Apr 2013 00:18:46 +0300</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
            <td>Yaron Sheffer <a class="moz-txt-link-rfc2396E" href="mailto:yaronf.ietf@gmail.com">&lt;yaronf.ietf@gmail.com&gt;</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
            <td>IETF Security Directorate <a class="moz-txt-link-rfc2396E" href="mailto:secdir@ietf.org">&lt;secdir@ietf.org&gt;</a>, The
              IESG <a class="moz-txt-link-rfc2396E" href="mailto:iesg@ietf.org">&lt;iesg@ietf.org&gt;</a>,
              <a class="moz-txt-link-abbreviated" href="mailto:draft-ietf-sipcore-sip-websocket.all@tools.ietf.org">draft-ietf-sipcore-sip-websocket.all@tools.ietf.org</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>I have reviewed this document as part of the security directorate's
ongoing effort to review all IETF documents being processed by the IESG.
These comments were written primarily for the benefit of the security
area directors.  Document editors and WG chairs should treat these
comments just like any other last call comments.

This document defines how SIP can be layered on the WebSocket protocol. 
This is essentially a profile of SIP.

Summary

In my opinion, from a security perspective this document is not yet 
ready for publication. I expect this document to be widely implemented, 
and I would not want it implemented until it offers a solid security 
model, even if it is optional.

Details

RFC 3261 has 20 pages of security considerations, and defines several 
security mechanisms. The current document "only" defines a transport for 
SIP, but such transport needs to preserve the security of any SIP 
mechanisms being used. In particular, it should be clear how endpoint 
identities are determined at each layer and how identities at different 
layers relate to one another. In other words, what's known as "channel 
binding".

The only security-related section (other than the Security 
Considerations) is Authentication (Sec. 7), and this section is marked 
non-normative. So the reader is left to wonder: how is the SIP client 
authenticated? How does this authentication relate to the 
WebSocket-level client authentication? And similarly for the server.

Sec. 9.1 mentions (again, non-normatively) that SIP-level certificate 
checks are omitted, and replaced by transport-level checks only. I 
believe this significantly reduces the security properties of SIP. 
Moreover, it begs for a policy tying WebSocket certificates to 
identities of the endpoints of the SIP protocol.

Nit/bug: Sec. 5.2.1 ends with a colon in the HTML version *only*, and is 
missing a paragraph.

Thanks,
	Yaron
</pre>
      <br>
    </div>
    <br>
  </body>
</html>

--------------060903000700040609090305--

From pkyzivat@alum.mit.edu  Thu Apr 11 09:18:54 2013
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Date: Fri, 12 Apr 2013 00:18:39 +0800
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: Richard Barnes <rlb@ipv.sx>, sipcore@ietf.org
Subject: Re: [sipcore] Fwd: SecDir review of draft-ietf-sipcore-sip-websocket-08
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 From first reading of the comments, my take is that this is going to 
require new normative text in the document. It isn't obvious to me what 
that text should be, but I suspect it is non-trivial. I think it will 
require mapping the http/https security models onto the sip security model.

I'll also note that since this is the first of several "X over 
Websockets" drafts, that it may set the pattern for what each of those 
will need to do.

I don't feel like I am strong enough on security issues to do this well. 
It would be great if we can find a volunteer strong on both SIP and 
security.

	Thanks,
	Paul

On 4/11/13 5:29 AM, Adam Roach wrote:
> A representative from the security directorate has weighed in on his
> opinion of the SIP over WebSockets document, and believes we may have
> additional work to do. I plan to spend some time wading through the
> specific comments and determining which ones indicate a need to change
> the document versus that that merely require an explanation for Yaron's
> benefit, but it will be a while before I have time to do so.
>
> If anyone else is in a position to perform that analysis and propose
> explanations and/or proposed document edits, please feel free to hop in
> and do so.
>
> /a
>
>
> -------- Original Message --------
> Subject: 	SecDir review of draft-ietf-sipcore-sip-websocket-08
> Resent-Date: 	Wed, 10 Apr 2013 16:19:04 -0500 (CDT)
> Resent-From: 	draft-alias-bounces@tools.ietf.org
> Resent-To: 	adam@nostrum.com, ibc@aliax.net, jmillan@aliax.net,
> pkyzivat@alum.mit.edu, rlb@ipv.sx, vpascual@acmepacket.com
> Date: 	Thu, 11 Apr 2013 00:18:46 +0300
> From: 	Yaron Sheffer <yaronf.ietf@gmail.com>
> To: 	IETF Security Directorate <secdir@ietf.org>, The IESG
> <iesg@ietf.org>, draft-ietf-sipcore-sip-websocket.all@tools.ietf.org
>
>
>
> I have reviewed this document as part of the security directorate's
> ongoing effort to review all IETF documents being processed by the IESG.
> These comments were written primarily for the benefit of the security
> area directors.  Document editors and WG chairs should treat these
> comments just like any other last call comments.
>
> This document defines how SIP can be layered on the WebSocket protocol.
> This is essentially a profile of SIP.
>
> Summary
>
> In my opinion, from a security perspective this document is not yet
> ready for publication. I expect this document to be widely implemented,
> and I would not want it implemented until it offers a solid security
> model, even if it is optional.
>
> Details
>
> RFC 3261 has 20 pages of security considerations, and defines several
> security mechanisms. The current document "only" defines a transport for
> SIP, but such transport needs to preserve the security of any SIP
> mechanisms being used. In particular, it should be clear how endpoint
> identities are determined at each layer and how identities at different
> layers relate to one another. In other words, what's known as "channel
> binding".
>
> The only security-related section (other than the Security
> Considerations) is Authentication (Sec. 7), and this section is marked
> non-normative. So the reader is left to wonder: how is the SIP client
> authenticated? How does this authentication relate to the
> WebSocket-level client authentication? And similarly for the server.
>
> Sec. 9.1 mentions (again, non-normatively) that SIP-level certificate
> checks are omitted, and replaced by transport-level checks only. I
> believe this significantly reduces the security properties of SIP.
> Moreover, it begs for a policy tying WebSocket certificates to
> identities of the endpoints of the SIP protocol.
>
> Nit/bug: Sec. 5.2.1 ends with a colon in the HTML version *only*, and is
> missing a paragraph.
>
> Thanks,
> 	Yaron
>
>
>
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


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From: Ofer Goren <oferg@avaya.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: Question regarding section 8.1.2 in 3261
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--_000_D9AD3D1627F52341B829F32E6DDE089149E5623Crvilmail2010_
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Hi All.
I have a question regarding section 8.1.2 in the 3261 (SIP) RFC.
The first paragraph says the following:
   The destination for the request is then computed.  Unless there is
   local policy specifying otherwise, the destination MUST be determined
   by applying the DNS procedures described in [4] as follows.  If the
   first element in the route set indicated a strict router (resulting
   in forming the request as described in Section 12.2.1.1), the
   procedures MUST be applied to the Request-URI of the request.
   Otherwise, the procedures are applied to the first Route header field
   value in the request (if one exists), or to the request's Request-URI
   if there is no Route header field present.  These procedures yield an
   ordered set of address, port, and transports to attempt.  Independent
   of which URI is used as input to the procedures of [4], if the
   Request-URI specifies a SIPS resource, the UAC MUST follow the
   procedures of [4] as if the input URI were a SIPS URI.

According to this, suppose I have client A calls client B using UDP, throug=
h a proxy. Client B answers 200OK with SIPS address in its CONTACT header. =
The proxy forwards the 200OK back to A, adding itself to the route set usin=
g record-route header, with "lr" parameter.
The ACK will be sent according to this: the request URI will be B's address=
, and the ROUTE header will include the address or the proxy.
The RFC states that regardless of the address of the proxy, the ACK should =
be sent using TLS transport, because B's address is SIPS.

What is the logic behind this? What if the proxy does not support TLS? Shou=
ldn't security be hop to hop?

Thanks,
Ofer

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<div class=3D"WordSection1">
<p class=3D"MsoNormal">Hi All.<o:p></o:p></p>
<p class=3D"MsoNormal">I have a question regarding section 8.1.2 in the 326=
1 (SIP) RFC.<o:p></o:p></p>
<p class=3D"MsoNormal">The first paragraph says the following:<o:p></o:p></=
p>
<p class=3D"MsoNormal">&nbsp;&nbsp; The destination for the request is then=
 computed.&nbsp; Unless there is<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; local policy specifying otherwise, the =
destination MUST be determined<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; by applying the DNS procedures describe=
d in [4] as follows.&nbsp; If the<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; first element in the route set indicate=
d a strict router (resulting<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; in forming the request as described in =
Section 12.2.1.1), the<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; procedures MUST be applied to the Reque=
st-URI of the request.<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; Otherwise, the procedures are applied t=
o the first Route header field<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; value in the request (if one exists), o=
r to the request's Request-URI<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; if there is no Route header field prese=
nt.&nbsp; These procedures yield an<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; ordered set of address, port, and trans=
ports to attempt. <b>
<u>&nbsp;Independent<o:p></o:p></u></b></p>
<p class=3D"MsoNormal"><b><u>&nbsp;&nbsp; of which URI is used as input to =
the procedures of [4], if the<o:p></o:p></u></b></p>
<p class=3D"MsoNormal"><b><u>&nbsp;&nbsp; Request-URI specifies a SIPS reso=
urce, the UAC MUST follow the<o:p></o:p></u></b></p>
<p class=3D"MsoNormal"><b><u>&nbsp;&nbsp; procedures of [4] as if the input=
 URI were a SIPS URI.<o:p></o:p></u></b></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">According to this, suppose I have client A calls cli=
ent B using UDP, through a proxy. Client B answers 200OK with SIPS address =
in its CONTACT header. The proxy forwards the 200OK back to A, adding itsel=
f to the route set using record-route
 header, with &#8220;lr&#8221; parameter.<o:p></o:p></p>
<p class=3D"MsoNormal">The ACK will be sent according to this: the request =
URI will be B&#8217;s address, and the ROUTE header will include the addres=
s or the proxy.
<o:p></o:p></p>
<p class=3D"MsoNormal"><b>The RFC states that regardless of the address of =
the proxy, the ACK should be sent using TLS transport, because B&#8217;s ad=
dress is SIPS.<o:p></o:p></b></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">What is the logic behind this? What if the proxy doe=
s not support TLS? Shouldn&#8217;t security be hop to hop?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thanks,<o:p></o:p></p>
<p class=3D"MsoNormal">Ofer<o:p></o:p></p>
</div>
</body>
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From: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
To: Ofer Goren <oferg@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: Question regarding section 8.1.2 in 3261
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Hi Oren,

RFC5630, which updated RFC3261, defines the proper handling of SIPS URI Sch=
eme.
Take a look at FC5630, section 4.1 for an answer to your question below.

Regards,
Rifaat


From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Ofer Goren
Sent: Wednesday, April 24, 2013 6:19 AM
To: sipcore@ietf.org
Subject: [sipcore] Question regarding section 8.1.2 in 3261

Hi All.
I have a question regarding section 8.1.2 in the 3261 (SIP) RFC.
The first paragraph says the following:
   The destination for the request is then computed.  Unless there is
   local policy specifying otherwise, the destination MUST be determined
   by applying the DNS procedures described in [4] as follows.  If the
   first element in the route set indicated a strict router (resulting
   in forming the request as described in Section 12.2.1.1), the
   procedures MUST be applied to the Request-URI of the request.
   Otherwise, the procedures are applied to the first Route header field
   value in the request (if one exists), or to the request's Request-URI
   if there is no Route header field present.  These procedures yield an
   ordered set of address, port, and transports to attempt.  Independent
   of which URI is used as input to the procedures of [4], if the
   Request-URI specifies a SIPS resource, the UAC MUST follow the
   procedures of [4] as if the input URI were a SIPS URI.

According to this, suppose I have client A calls client B using UDP, throug=
h a proxy. Client B answers 200OK with SIPS address in its CONTACT header. =
The proxy forwards the 200OK back to A, adding itself to the route set usin=
g record-route header, with "lr" parameter.
The ACK will be sent according to this: the request URI will be B's address=
, and the ROUTE header will include the address or the proxy.
The RFC states that regardless of the address of the proxy, the ACK should =
be sent using TLS transport, because B's address is SIPS.

What is the logic behind this? What if the proxy does not support TLS? Shou=
ldn't security be hop to hop?

Thanks,
Ofer

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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Hi Oren,<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">RFC5630, which updated=
 RFC3261, defines the proper handling of SIPS URI Scheme.<o:p></o:p></span>=
</p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Take a look at FC5630,=
 section 4.1 for an answer to your question below.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Regards,<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Rifaat<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> sipcore-=
bounces@ietf.org [mailto:sipcore-bounces@ietf.org]
<b>On Behalf Of </b>Ofer Goren<br>
<b>Sent:</b> Wednesday, April 24, 2013 6:19 AM<br>
<b>To:</b> sipcore@ietf.org<br>
<b>Subject:</b> [sipcore] Question regarding section 8.1.2 in 3261<o:p></o:=
p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Hi All.<o:p></o:p></p>
<p class=3D"MsoNormal">I have a question regarding section 8.1.2 in the 326=
1 (SIP) RFC.<o:p></o:p></p>
<p class=3D"MsoNormal">The first paragraph says the following:<o:p></o:p></=
p>
<p class=3D"MsoNormal">&nbsp;&nbsp; The destination for the request is then=
 computed.&nbsp; Unless there is<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; local policy specifying otherwise, the =
destination MUST be determined<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; by applying the DNS procedures describe=
d in [4] as follows.&nbsp; If the<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; first element in the route set indicate=
d a strict router (resulting<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; in forming the request as described in =
Section 12.2.1.1), the<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; procedures MUST be applied to the Reque=
st-URI of the request.<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; Otherwise, the procedures are applied t=
o the first Route header field<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; value in the request (if one exists), o=
r to the request's Request-URI<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; if there is no Route header field prese=
nt.&nbsp; These procedures yield an<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; ordered set of address, port, and trans=
ports to attempt. <b>
<u>&nbsp;Independent<o:p></o:p></u></b></p>
<p class=3D"MsoNormal"><b><u>&nbsp;&nbsp; of which URI is used as input to =
the procedures of [4], if the<o:p></o:p></u></b></p>
<p class=3D"MsoNormal"><b><u>&nbsp;&nbsp; Request-URI specifies a SIPS reso=
urce, the UAC MUST follow the<o:p></o:p></u></b></p>
<p class=3D"MsoNormal"><b><u>&nbsp;&nbsp; procedures of [4] as if the input=
 URI were a SIPS URI.<o:p></o:p></u></b></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">According to this, suppose I have client A calls cli=
ent B using UDP, through a proxy. Client B answers 200OK with SIPS address =
in its CONTACT header. The proxy forwards the 200OK back to A, adding itsel=
f to the route set using record-route
 header, with &#8220;lr&#8221; parameter.<o:p></o:p></p>
<p class=3D"MsoNormal">The ACK will be sent according to this: the request =
URI will be B&#8217;s address, and the ROUTE header will include the addres=
s or the proxy.
<o:p></o:p></p>
<p class=3D"MsoNormal"><b>The RFC states that regardless of the address of =
the proxy, the ACK should be sent using TLS transport, because B&#8217;s ad=
dress is SIPS.<o:p></o:p></b></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">What is the logic behind this? What if the proxy doe=
s not support TLS? Shouldn&#8217;t security be hop to hop?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thanks,<o:p></o:p></p>
<p class=3D"MsoNormal">Ofer<o:p></o:p></p>
</div>
</div>
</body>
</html>

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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Ofer Goren <oferg@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: Question regarding section 8.1.2 in 3261
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You may want to read RFC 5630 as it updates some of the SIPs usage in RFC 3=
261, but also explains the justification behind some of the usage.

Keith

________________________________
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Ofer Goren
Sent: 24 April 2013 11:19
To: sipcore@ietf.org
Subject: [sipcore] Question regarding section 8.1.2 in 3261

Hi All.
I have a question regarding section 8.1.2 in the 3261 (SIP) RFC.
The first paragraph says the following:
   The destination for the request is then computed.  Unless there is
   local policy specifying otherwise, the destination MUST be determined
   by applying the DNS procedures described in [4] as follows.  If the
   first element in the route set indicated a strict router (resulting
   in forming the request as described in Section 12.2.1.1), the
   procedures MUST be applied to the Request-URI of the request.
   Otherwise, the procedures are applied to the first Route header field
   value in the request (if one exists), or to the request's Request-URI
   if there is no Route header field present.  These procedures yield an
   ordered set of address, port, and transports to attempt.  Independent
   of which URI is used as input to the procedures of [4], if the
   Request-URI specifies a SIPS resource, the UAC MUST follow the
   procedures of [4] as if the input URI were a SIPS URI.

According to this, suppose I have client A calls client B using UDP, throug=
h a proxy. Client B answers 200OK with SIPS address in its CONTACT header. =
The proxy forwards the 200OK back to A, adding itself to the route set usin=
g record-route header, with "lr" parameter.
The ACK will be sent according to this: the request URI will be B's address=
, and the ROUTE header will include the address or the proxy.
The RFC states that regardless of the address of the proxy, the ACK should =
be sent using TLS transport, because B's address is SIPS.

What is the logic behind this? What if the proxy does not support TLS? Shou=
ldn't security be hop to hop?

Thanks,
Ofer

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<div class=3D"Section1">
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">You may want to read RFC 5630 as it up=
dates some of the SIPs usage in RFC 3261, but also explains the justificati=
on behind some of the
 usage.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">Keith<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt">
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><font=
 size=3D"3" face=3D"Times New Roman"><span lang=3D"EN-US" style=3D"font-siz=
e:12.0pt;font-family:
&quot;Times New Roman&quot;">
<hr size=3D"2" width=3D"100%" align=3D"center" tabindex=3D"-1">
</span></font></div>
<p class=3D"MsoNormal"><b><font size=3D"2" face=3D"Tahoma"><span lang=3D"EN=
-US" style=3D"font-size:10.0pt;font-family:Tahoma;font-weight:bold">From:</=
span></font></b><font size=3D"2" face=3D"Tahoma"><span lang=3D"EN-US" style=
=3D"font-size:10.0pt;font-family:Tahoma"> sipcore-bounces@ietf.org
 [mailto:sipcore-bounces@ietf.org] <b><span style=3D"font-weight:bold">On B=
ehalf Of
</span></b>Ofer Goren<br>
<b><span style=3D"font-weight:bold">Sent:</span></b> 24 April 2013 11:19<br=
>
<b><span style=3D"font-weight:bold">To:</span></b> sipcore@ietf.org<br>
<b><span style=3D"font-weight:bold">Subject:</span></b> [sipcore] Question =
regarding section 8.1.2 in 3261</span></font><font size=3D"3" face=3D"Times=
 New Roman"><span lang=3D"EN-US" style=3D"font-size:12.0pt;font-family:&quo=
t;Times New Roman&quot;"><o:p></o:p></span></font></p>
</div>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span style=3D"fon=
t-size:11.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">Hi All.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">I have a question regarding section 8.1.2 in the 3261 (SIP) RFC.<o:=
p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">The first paragraph says the following:<o:p></o:p></span></font></p=
>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">&nbsp;&nbsp; The destination for the request is then computed.&nbsp=
; Unless there is<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">&nbsp;&nbsp; local policy specifying otherwise, the destination MUS=
T be determined<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">&nbsp;&nbsp; by applying the DNS procedures described in [4] as fol=
lows.&nbsp; If the<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">&nbsp;&nbsp; first element in the route set indicated a strict rout=
er (resulting<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">&nbsp;&nbsp; in forming the request as described in Section 12.2.1.=
1), the<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">&nbsp;&nbsp; procedures MUST be applied to the Request-<st1:stockti=
cker w:st=3D"on">URI</st1:stockticker> of the request.<o:p></o:p></span></f=
ont></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">&nbsp;&nbsp; Otherwise, the procedures are applied to the
<st1:Street w:st=3D"on"><st1:address w:st=3D"on">first Route</st1:address><=
/st1:Street> header field<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">&nbsp;&nbsp; value in the request (if one exists), or to the reques=
t's Request-<st1:stockticker w:st=3D"on">URI</st1:stockticker><o:p></o:p></=
span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">&nbsp;&nbsp; if there is no Route header field present.&nbsp; These=
 procedures yield an<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">&nbsp;&nbsp; ordered set of address, port, and transports to attemp=
t.
<b><u><span style=3D"font-weight:bold">&nbsp;Independent<o:p></o:p></span><=
/u></b></span></font></p>
<p class=3D"MsoNormal"><b><u><font size=3D"2" face=3D"Calibri"><span lang=
=3D"EN-US" style=3D"font-size:11.0pt;font-weight:bold">&nbsp;&nbsp; of whic=
h
<st1:stockticker w:st=3D"on">URI</st1:stockticker> is used as input to the =
procedures of [4], if the<o:p></o:p></span></font></u></b></p>
<p class=3D"MsoNormal"><b><u><font size=3D"2" face=3D"Calibri"><span lang=
=3D"EN-US" style=3D"font-size:11.0pt;font-weight:bold">&nbsp;&nbsp; Request=
-<st1:stockticker w:st=3D"on">URI</st1:stockticker> specifies a SIPS resour=
ce, the UAC MUST follow the<o:p></o:p></span></font></u></b></p>
<p class=3D"MsoNormal"><b><u><font size=3D"2" face=3D"Calibri"><span lang=
=3D"EN-US" style=3D"font-size:11.0pt;font-weight:bold">&nbsp;&nbsp; procedu=
res of [4] as if the input
<st1:stockticker w:st=3D"on">URI</st1:stockticker> were a SIPS <st1:stockti=
cker w:st=3D"on">
URI</st1:stockticker>.<o:p></o:p></span></font></u></b></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">According to this, suppose I have client A calls client B using UDP=
, through a proxy. Client B answers 200OK with SIPS address in its CONTACT =
header. The proxy
 forwards the 200OK back to A, adding itself to the route set using record-=
route header, with &#8220;lr&#8221; parameter.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">The ACK will be sent according to this: the request
<st1:stockticker w:st=3D"on">URI</st1:stockticker> will be B&#8217;s addres=
s, and the ROUTE header will include the address or the proxy.
<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><b><font size=3D"2" face=3D"Calibri"><span lang=3D"E=
N-US" style=3D"font-size:11.0pt;font-weight:bold">The RFC states that regar=
dless of the address of the proxy, the ACK should be sent using
<st1:stockticker w:st=3D"on">TLS</st1:stockticker> transport, because B&#82=
17;s address is SIPS.<o:p></o:p></span></font></b></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">What is the logic behind this? What if the proxy does not support
<st1:stockticker w:st=3D"on">TLS</st1:stockticker>? Shouldn&#8217;t securit=
y be hop to hop?<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">Thanks,<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Calibri"><span lang=3D"EN-U=
S" style=3D"font-size:
11.0pt">Ofer<o:p></o:p></span></font></p>
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--1466435115-244288905-1366891827=:70197
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=0A=0ADear Experts,=A0=0A=A0 =A0 =A0 =A0 =A0 =A0I am having a following cas=
e, where the REGISTER request To header URI and the Contact URI are same. P=
ls see the message below.=0A=0AREGISTER sip:10.232.15.206:5070 SIP/2.0=0ATo=
: <sip:1001@10.232.15.206>=0AVia: SIP/2.0/TCP 10.232.122.70;branch=3Dz9hG4b=
K-6928-1-0=0AFrom: <sip:1001@10.232.15.206>;tag=3D123451001=0ACall-ID: 1-69=
28@10.232.122.70=0ACSeq: 11 REGISTER=0A=0AMax-Forwards: 70=0A=0Asupported: =
gruu,path,outbound=0A=0Arequire: gruu=0A=0AContact: <sip:1001@10.232.15.206=
;deviceName=3Ddevice1>;+sip.instance=3D"<urn:uuid:f81d4fae-7dec-11d0-a765-0=
0a0c9e6bf6>"=0A=0AContent-Length: 0=0A=0A=0ARFC 5627 says,=A0=0AThe contact=
 URI MUST NOT be equivalent, based on=0A=A0 =A0the URI equality rules in RF=
C 3261 [1], to the AOR in the To header=0A=A0 =A0field.=A0If the contact UR=
I is equivalent (based on URI equivalence in RFC=0A=A0 =A03261 [1]) to the =
AOR, the registrar MUST reject the request with a=0A=A0 =A0403, since this =
would cause a routing loop.=A0=0A=0AFrom RFC 3261, URI equality rules say,=
=A0=0AURI uri-parameter components are compared as follows:=0A=A0 =A0 =A0 =
=A0 =A0- =A0Any uri-parameter appearing in both URIs must match.=0A=A0 =A0 =
=A0 =A0 =A0- =A0A user, ttl, or method uri-parameter appearing in only one=
=0A=0A=A0 =A0 =A0 =A0 =A0 =A0 URI never matches, even if it contains the de=
fault value.=0A=A0 =A0 =A0 =A0 =A0- =A0A URI that includes an maddr paramet=
er will not match a URI=0A=0A=A0 =A0 =A0 =A0 =A0 =A0 that contains no maddr=
 parameter.=0A=A0 =A0 =A0 =A0 =A0- =A0All other uri-parameters appearing in=
 only one URI are=0A=0A=A0 =A0 =A0 =A0 =A0 =A0 ignored when comparing the U=
RIs.=0A=A0URI header components are never ignored. =A0Any present header=0A=
=0A=A0 =A0 =A0 =A0 =A0component MUST be present in both URIs and match for =
the URIs=0A=A0 =A0 =A0 =A0 =A0to match. =A0The matching rules are defined f=
or each header field=0A=A0 =A0 =A0 =A0 =A0in Section 20.=0A=0ANow, while ch=
ecking To and Contact URIs for equality, we can ignore the uncommon URI par=
ameters in the Contact URI(here it is deviceName). But do we need to consid=
er the Contact "header" parameters(here in above message +sip.instance para=
m) also while comparing the URIs in To and Contact to decide that they are =
equal and reject the request with 403 as it would cause loop during lookup =
? Getting this doubt bcos RFC 3261 says "URI header components are never ig=
nored". What is an URI header component ? Does it apply to Contact header p=
arameter in this case ?=0A=0A=0AIn this case, is the above REGISTER request=
 to be rejected with 403 or not ?=0A=0Aregards=0A+Santhana
--1466435115-244288905-1366891827=:70197
Content-Type: text/html; charset=iso-8859-1
Content-Transfer-Encoding: quoted-printable

<html><body><div style=3D"color:#000; background-color:#fff; font-family:ti=
mes new roman, new york, times, serif;font-size:12pt"><div style=3D"font-fa=
mily: 'times new roman', 'new york', times, serif; font-size: 12pt;"><br></=
div><div style=3D"font-family: 'times new roman', 'new york', times, serif;=
 font-size: 16px; color: rgb(0, 0, 0); background-color: transparent; font-=
style: normal;">Dear Experts,&nbsp;</div><div style=3D"font-family: 'times =
new roman', 'new york', times, serif; font-size: 16px; color: rgb(0, 0, 0);=
 background-color: transparent; font-style: normal;">&nbsp; &nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp;I am having a following case, where the REGISTER request=
 To header URI and the Contact URI are same. Pls see the message below.</di=
v><div style=3D"font-family: 'times new roman', 'new york', times, serif; f=
ont-size: 16px; color: rgb(0, 0, 0); background-color: transparent; font-st=
yle: normal;"><br></div><div style=3D"background-color: transparent;"><div
 style=3D"background-color: transparent;"><span style=3D"font-family: 'time=
s new roman', 'new york', times, serif; background-color: transparent;">REG=
ISTER sip:10.232.15.206:5070 SIP/2.0</span></div><div style=3D"background-c=
olor: transparent;"><font face=3D"times new roman, new york, times, serif">=
To: &lt;sip:1001@10.232.15.206&gt;</font></div><div style=3D"background-col=
or: transparent;"><font face=3D"times new roman, new york, times, serif">Vi=
a: SIP/2.0/TCP 10.232.122.70;branch=3Dz9hG4bK-6928-1-0</font></div><div sty=
le=3D"background-color: transparent;"><font face=3D"times new roman, new yo=
rk, times, serif">From: &lt;sip:1001@10.232.15.206&gt;;tag=3D123451001</fon=
t></div><div style=3D"background-color: transparent;"><font face=3D"times n=
ew roman, new york, times, serif">Call-ID: 1-6928@10.232.122.70</font></div=
><div style=3D"background-color: transparent;"><span style=3D"font-family: =
'times new roman', 'new york', times, serif; background-color: transparent;=
">CSeq: 11
 REGISTER</span><br></div><div style=3D"background-color: transparent;"><sp=
an style=3D"background-color: transparent; font-family: 'times new roman', =
'new york', times, serif;">Max-Forwards: 70</span><br></div><div style=3D"b=
ackground-color: transparent;"><span style=3D"font-family: 'times new roman=
', 'new york', times, serif; background-color: transparent;">supported: gru=
u,path,outbound</span><br></div><div style=3D"background-color: transparent=
;"><span style=3D"font-family: 'times new roman', 'new york', times, serif;=
 background-color: transparent;">require: gruu</span><br></div><div style=
=3D"background-color: transparent;"><span style=3D"font-family: 'times new =
roman', 'new york', times, serif; background-color: transparent;">Contact: =
&lt;sip:1001@10.232.15.206;deviceName=3Ddevice1&gt;;+sip.instance=3D"&lt;ur=
n:uuid:f81d4fae-7dec-11d0-a765-00a0c9e6bf6&gt;"</span><br></div><div style=
=3D"background-color: transparent;"><span style=3D"font-family: 'times new =
roman', 'new york',
 times, serif; background-color: transparent;">Content-Length: 0</span><br>=
</div><div style=3D"background-color: transparent; color: rgb(0, 0, 0); fon=
t-size: 16px; font-family: 'Times New Roman'; font-style: normal;"><span st=
yle=3D"font-family: 'times new roman', 'new york', times, serif; background=
-color: transparent;"><br></span></div><div style=3D"background-color: tran=
sparent; color: rgb(0, 0, 0); font-size: 16px; font-family: 'times new roma=
n', 'new york', times, serif; font-style: normal;"><span style=3D"font-fami=
ly: 'times new roman', 'new york', times, serif; background-color: transpar=
ent;">RFC 5627 says,&nbsp;</span></div><div style=3D"background-color: tran=
sparent;"><span style=3D"background-color: transparent;"><font face=3D"time=
s new roman, new york, times, serif"><div style=3D"background-color: transp=
arent;">The contact URI MUST NOT be equivalent, based on</div><div style=3D=
"background-color: transparent;">&nbsp; &nbsp;the URI equality rules in RFC=
 3261 [1],
 to the AOR in the To header</div><div style=3D"background-color: transpare=
nt;">&nbsp; &nbsp;field.&nbsp;<span style=3D"background-color: transparent;=
">If the contact URI is equivalent (based on URI equivalence in RFC</span><=
/div><div style=3D"background-color: transparent;">&nbsp; &nbsp;3261 [1]) t=
o the AOR, the registrar MUST reject the request with a</div><div style=3D"=
background-color: transparent;">&nbsp; &nbsp;403, since this would cause a =
routing loop.&nbsp;</div><div style=3D"background-color: transparent;"><br>=
</div><div style=3D"background-color: transparent; color: rgb(0, 0, 0); fon=
t-size: 16px; font-family: 'times new roman', 'new york', times, serif; fon=
t-style: normal;">From RFC 3261, URI equality rules say,&nbsp;</div><div st=
yle=3D"background-color: transparent;"><div style=3D"background-color: tran=
sparent;">URI uri-parameter components are compared as follows:</div><div s=
tyle=3D"background-color: transparent;">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;-=
 &nbsp;Any
 uri-parameter appearing in both URIs must match.</div><div style=3D"backgr=
ound-color: transparent;"><span style=3D"background-color: transparent;">&n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp;- &nbsp;A user, ttl, or method uri-paramete=
r appearing in only one</span><br></div><div style=3D"background-color: tra=
nsparent;">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; URI never matches, eve=
n if it contains the default value.</div><div style=3D"background-color: tr=
ansparent;"><span style=3D"background-color: transparent;">&nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp;- &nbsp;A URI that includes an maddr parameter will not m=
atch a URI</span><br></div><div style=3D"background-color: transparent;">&n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; that contains no maddr parameter.</=
div><div style=3D"background-color: transparent;"><span style=3D"background=
-color: transparent;">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;- &nbsp;All other u=
ri-parameters appearing in only one URI are</span><br></div><div
 style=3D"background-color: transparent;">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
; &nbsp; ignored when comparing the URIs.</div><div style=3D"background-col=
or: transparent;"><span style=3D"background-color: transparent;">&nbsp;URI =
header components are never ignored. &nbsp;Any present header</span><br></d=
iv><div style=3D"background-color: transparent;">&nbsp; &nbsp; &nbsp; &nbsp=
; &nbsp;component MUST be present in both URIs and match for the URIs</div>=
<div style=3D"background-color: transparent;">&nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp;to match. &nbsp;The matching rules are defined for each header field</=
div><div style=3D"background-color: transparent;">&nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp;in Section 20.</div><div style=3D"color: rgb(0, 0, 0); font-family=
: 'times new roman', 'new york', times, serif; font-size: 16px; font-style:=
 normal;"><br></div><div style=3D"color: rgb(0, 0, 0); font-family: 'times =
new roman', 'new york', times, serif; font-size: 16px; font-style: normal;"=
><span
 style=3D"background-color: transparent;">Now, while checking To and Contac=
t URIs for equality, we can ignore the uncommon URI parameters in the Conta=
ct URI(here it is deviceName). But do we need to consider the Contact "head=
er" parameters(here in above message +sip.instance param) also while compar=
ing the URIs in To and Contact to decide that they are equal and reject the=
 request with 403 as it would cause loop during lookup ? Getting this doubt=
 bcos RFC 3261 says "URI header components are never ignored". What is an U=
RI header component ? Does it apply to Contact header parameter in this cas=
e ?</span><br></div><div style=3D"color: rgb(0, 0, 0); font-family: 'times =
new roman', 'new york', times, serif; font-size: 16px; font-style: normal;"=
><br></div><div style=3D"color: rgb(0, 0, 0); font-family: 'times new roman=
', 'new york', times, serif; font-size: 16px; font-style: normal;">In this =
case, is the above REGISTER request to be rejected with 403 or not
 ?</div><div style=3D"color: rgb(0, 0, 0); font-family: 'times new roman', =
'new york', times, serif; font-size: 16px; font-style: normal;"><br></div><=
div style=3D"color: rgb(0, 0, 0); font-family: 'times new roman', 'new york=
', times, serif; font-size: 16px; font-style: normal;">regards</div><div st=
yle=3D"color: rgb(0, 0, 0); font-family: 'times new roman', 'new york', tim=
es, serif; font-size: 16px; font-style: normal;">+Santhana</div></div></fon=
t></span></div></div></div></body></html>
--1466435115-244288905-1366891827=:70197--

From worley@shell01.TheWorld.com  Thu Apr 25 10:26:08 2013
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From: worley@ariadne.com (Dale R. Worley)
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To: Santhana Krishnan <yskrishnan@yahoo.com>
In-reply-to: <1366891827.70197.YahooMailNeo@web160403.mail.bf1.yahoo.com> (yskrishnan@yahoo.com)
References: <1366891827.70197.YahooMailNeo@web160403.mail.bf1.yahoo.com>
Cc: sipcore@ietf.org
Subject: Re: [sipcore] REGISTER AOR Loop Scenario.
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> From: Santhana Krishnan <yskrishnan@yahoo.com>

> To: <sip:1001@10.232.15.206>

> Contact: <sip:1001@10.232.15.206;deviceName=device1>;+sip.instance="<urn:uuid:f81d4fae-7dec-11d0-a765-00a0c9e6bf6>"

> Now, while checking To and Contact URIs for equality, we can ignore
> the uncommon URI parameters in the Contact URI(here it is
> deviceName). But do we need to consider the Contact "header"
> parameters(here in above message +sip.instance param) also while
> comparing the URIs in To and Contact to decide that they are equal
> and reject the request with 403 as it would cause loop during lookup
> ? Getting this doubt bcos RFC 3261 says "URI header components are
> never ignored". What is an URI header component ? Does it apply to
> Contact header parameter in this case ?

The "URI" in the To header is "sip:1001@10.232.15.206".
The "URI" in the Contact header is "sip:1001@10.232.15.206;deviceName=device1".

The "+sip.instance=..." part is *not* part of the URI, it is a
"contact-param" -- see the grammar rule "Contact" in section 25.1 of
RFC 3261.  These sort of parameters attached to URIs in the values of
header fields are often called "field parameters" because they have
the same syntax in many different header fields, although the grammar
in 25.1 uses different non-terminal names depending on which header
field they appear in.

The "URI header components" referred to in section 19.1.4 refers to
the nonterminal "headers" in section 25.1, that is, constructs like
"?Subject=foo" in the URI "sip:user@host?Subject=foo".

Dale

From hammondjohnson@hushmail.com  Sat Apr 27 14:06:57 2013
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Subject: [sipcore] Biggest Fake Conference in Computer Science
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We are researchers from different parts of the world and conducted a study on  
the world’s biggest bogus computer science conference WORLDCOMP 
( http://sites.google.com/site/worlddump1 ) organized by Prof. Hamid Arabnia 
from University of Georgia, USA.


We submitted a fake paper to WORLDCOMP 2011 and again (the same paper 
with a modified title) to WORLDCOMP 2012. This paper had numerous 
fundamental mistakes. Sample statements from that paper include: 

(1). Binary logic is fuzzy logic and vice versa
(2). Pascal developed fuzzy logic
(3). Object oriented languages do not exhibit any polymorphism or inheritance
(4). TCP and IP are synonyms and are part of OSI model 
(5). Distributed systems deal with only one computer
(6). Laptop is an example for a super computer
(7). Operating system is an example for computer hardware


Also, our paper did not express any conceptual meaning.  However, it 
was accepted both the times without any modifications (and without 
any reviews) and we were invited to submit the final paper and a 
payment of $500+ fee to present the paper. We decided to use the 
fee for better purposes than making Prof. Hamid Arabnia (Chairman 
of WORLDCOMP) rich. After that, we received few reminders from 
WORLDCOMP to pay the fee but we never responded. 


We MUST say that you should look at the above website if you have any thoughts 
to submit a paper to WORLDCOMP.  DBLP and other indexing agencies have stopped 
indexing WORLDCOMP’s proceedings since 2011 due to its fakeness. See 
http://www.informatik.uni-trier.de/~ley/db/conf/icai/index.html for of one of the 
conferences of WORLDCOMP and notice that there is no listing after 2010. See Section 2 of
http://sites.google.com/site/dumpconf for comments from well-known researchers 
about WORLDCOMP. 


The status of your WORLDCOMP papers can be changed from scientific
to other (i.e., junk or non-technical) at any time. Better not to have a paper than 
having it in WORLDCOMP and spoil the resume and peace of mind forever!


Our study revealed that WORLDCOMP is a money making business, 
using University of Georgia mask, for Prof. Hamid Arabnia. He is throwing 
out a small chunk of that money (around 20 dollars per paper published 
in WORLDCOMP’s proceedings) to his puppet (Mr. Ashu Solo or A.M.G. Solo) 
who publicizes WORLDCOMP and also defends it at various forums, using 
fake/anonymous names. The puppet uses fake names and defames other conferences
to divert traffic to WORLDCOMP. He also makes anonymous phone calls and tries to 
threaten the critiques of WORLDCOMP (See Item 7 of Section 5 of above website). 
That is, the puppet does all his best to get a maximum number of papers published 
at WORLDCOMP to get more money into his (and Prof. Hamid Arabnia’s) pockets. 


Monte Carlo Resort (the venue of WORLDCOMP for more than 10 years, until 2012) has 
refused to provide the venue for WORLDCOMP’13 because of the fears of their image 
being tarnished due to WORLDCOMP’s fraudulent activities. That is why WORLDCOMP’13 
is taking place at a different resort. WORLDCOMP will not be held after 2013. 


The draft paper submission deadline is over but still there are no committee 
members, no reviewers, and there is no conference Chairman. The only contact 
details available on WORLDCOMP’s website is just an email address! 

Let us make a direct request to Prof. Hamid arabnia: publish all reviews for 
all the papers (after blocking identifiable details) since 2000 conference. Reveal 
the names and affiliations of all the reviewers (for each year) and how many 
papers each reviewer had reviewed on average. We also request him to look at 
the Open Challenge (Section 6) at https://sites.google.com/site/moneycomp1 


Sorry for posting to multiple lists. Spreading the word is the only way to stop 
this bogus conference. Please forward this message to other mailing lists and people. 


We are shocked with Prof. Hamid Arabnia and his puppet’s activities 
http://worldcomp-fake-bogus.blogspot.com   Search Google using the 
keyword worldcomp fake for additional links.

