On Tue, Jul 12, 2011 at 10:06 AM, Iñaki Baz Castillo <i...@aliax.net> wrote:
> 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? > > Interesting thing, rfc3261, Section 8.1.1.8 says: If the Request-URI or top Route header field value contains a SIPS URI, the Contact header field MUST contain a SIPS URI as well. >From this, I understand that you can put SIP into RURI and SIPS into Contact when sending a request, but you should as well add a top Route header(the outbound proxy) having a SIPS schema. Regards, Brez _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
