1 Dec
2012
1 Dec
'12
12:29 a.m.
Andreas Granig writes:
A lot of clients and servers are sending
"sip:user@example.com;transport=tls" in request URIs and Contact headers
and Record-Route headers, and you can check with
uri_param("transport","tls") which transport socket to use. This is
pretty useful as you can determine hop-by-hop whether or not to use TLS.
This approach has been obsoleted by RFC3261 though, and there doesn't
seem to be a mechanism in RFC3263 to indicate "use schema sip, but use
transport=tls".
when rfc3261 was specified, the whole sips thing was put there at the
last minute. no-one had any practical experience of it. i guess the
requirement to have something like https came from ietf area directors.
the result is a useless mess.
And what's the general take on this
"sips" schema? As far as I
understand RFC3261, it means that if a client sends a request to a
sips-URI, the request is sent to the domain via TLS, and from there "the
request is sent securely to the callee, but with security mechanisms
that depend on the policy of the domain of the callee." (RFC3261,
Chapter 4). What does this really mean in practice? Are you allowed to
rewrite the schema to "sip" and pass it on for example via UDP to the
callee if the callee didn't indicate transport=tls (deprecated anyways)
or "sips:" in the Contact of the registration?
in that case, the proxy has to drop the request unless the last hop is
"secure", i.e., uses a vpn, barb wire, or something.
Or should you keep "sips"
as schema, but still send it via UDP, because you know based on local
policy or based on client registration that the next hop is not
supporting TLS? How would widespread clients react when getting a call
to a "sips" URI, especially if they receive it via UDP?
you cannot use sips with udp transport.
-- juha