Re: [Sip] Using TLS in the first hop - Bug in RFC 5630

Iñaki Baz Castillo <ibc@aliax.net> Thu, 15 September 2011 14:28 UTC

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Date: Thu, 15 Sep 2011 16:30:52 +0200
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From: Iñaki Baz Castillo <ibc@aliax.net>
To: "Horvath, Ernst" <ernst.horvath@siemens-enterprise.com>
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Cc: "sip@ietf.org" <sip@ietf.org>
Subject: Re: [Sip] Using TLS in the first hop - Bug in RFC 5630
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2011/9/15 Iñaki Baz Castillo <ibc@aliax.net>:
>> It may work for the 1st request. But in a subsequent mid-dialog request in the reverse direction the contact URI becomes the Request-URI, which is now SIPS, and therefore the Contact in this request must also become SIPS, and you end up in an all-SIPS case.

This is not true. The in-dialog request will have, indeed, a RURI with
SIPS schema, but it will also contain a top Route with SIP schema, so
it takes preference and there is no need at all to set a SIPS Contact
URI.

RFC 5630:

5.1.2.  UAS Behavior

   When presented with a SIPS URI, a UAS MUST NOT change it to a SIP
   URI.

   As mandated by [RFC3261], Section 12.1.1:

      If the request that initiated the dialog contained a SIPS URI in
      the Request-URI or in the top Record-Route header field value, if
      there was any, or the Contact header field if there was no Record-
      Route header field, the Contact header field in the response MUST
      be a SIPS URI.

In our case *there is* a Record-Route in the INVITE arriving to the
UAS and it does contain a SIP URI, so the Contact in the response from
the UAS does not need to be a SIPS URI.


Anyhow I insist: SIPS and TLS usage is a pain. Nobody understands it
(it's clear given any mail thread about this topic) and everybody does
wrong assumptions. I think SIP authors should worry about this reality
and react.


Regards.


-- 
Iñaki Baz Castillo
<ibc@aliax.net>