2011/7/11 Iñaki Baz Castillo <i...@aliax.net>: > But, if the UA (caller) uses a SIP URI in the Contact URI of the > INVITE, in-dialog requests from the callee would nor arrive to the > caller via TLS. This is: > > INVITE sip:b...@domain.com > Via: SIP/2.0/TLS 1.2.3.4 > Contact: <sip:alice@1.2.3.4> > > This means that when Bob sends a BYE it would arrive to Alice's > outbound proxy as follows: > > INVITE sip:alice@1.2.3.4 > Route: xxxxxxx > > The proxy would remove Route header(s) and just the RURI remains, > which has no SIPS scheme so it would send the request using UDP (or > TCP if the Contact URI includes a ;transport=tcp). > > Is it correct? Wouldn't be better that the Contact in the INVITE > contains a SIPS schema?
Hi, any comment about it please? :) Thanks. -- Iñaki Baz Castillo <i...@aliax.net> _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors