Why would you want to use sips in the Route header in this particular
case and not sip with the transport=tls parameter? I tried to make
sense out of rfc 5630 but failed, so if you could present any more
evidence why this is a good idea, that would be great.
The way I understand sips is that by using this scheme in a URI, you
are indicating that you want to be contacted securely and if that is
not possible (for whatever reason), you would rather want the request
to fail than to be forwarded insecurely. I can see how this could make
sense from end-point's (phone) point of view.
But why would anyone want to use this in the Route header? By using
sips instead of sip;transport=tls in a Route header the proxy server
is enforcing that it is to be contacted securely during the dialog,
even if the original request (that established the dialog) may not
have been subject to such a restriction. In other words request path
that was fine for the INVITE is no longer fine for the re-INVITE or
BYE.
Also, what is the real difference between using sip;transport=tls and
sips in Route headers? In both cases the server sending the request
will fail to deliver it if it cannot contact the downstream server
over TLS. Or is it supposed to fallback to UDP when the parameter is
used and reject the request when sips is used?
-Jan
On Tue, Jul 5, 2011 at 09:03, Iñaki Baz Castillo <ibc(a)aliax.net> wrote:
Hi, when sip-router (master branch) receives a request
via TLS (but
with a "sip:" scheme URI in the Request Line) and routes the request
using other transport (as UDP) it appends a Record-Route as follows:
Record-Route: <sip:IP_PROXY:PORT;transport=tls>
According to RFC 3261 and RFC 5630 (which explains better sips schema
usage) the correct Record-Route should be:
Record-Route: <sips:IP_PROXY:PORT>
or:
Record-Route: <sips:IP_PROXY:PORT;transport=tcp>
However Sip-Router behaves a bit deprecated by using ;transport=tls,
which is clearly deprecated even in RFC 3261.
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
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