On 14/01/13 15:59, Klaus Darilion wrote: > The caller should use the NATPR and thus should use TLS. The SIPS+D2T > does not requires the URI to be a SIPS URI. >
That was my understanding too - do you feel it is always working this way in practice though with the major SIP proxies/PBXes? Or are any extra efforts (such as NAPTR for rewriting sip: to sips:) needed to help non-conforming implementations? > See also the thread > "NAPTR, SRV and sips vs. transport=tls" from 1.Dec.2012 > Yes, I did see that previously but the focus of my question was slightly different, hence a new thread > regards > Klaus > > On 11.01.2013 18:45, Daniel Pocock wrote: >> >> >> >> I'm just wondering if anyone can comment on expected and actual behavior >> if there is only a NAPTR record for TLS, e.g. I have: >> >> sip5060.net. IN NAPTR 10 0 "s" "SIPS+D2T" "" >> _sips._tcp.sip5060.net. >> >> >> >> and I don't have any entry for "SIP+D2U" or "SIP+D2T" >> >> If some third party Kamailio instance (e.g. sip-server.example.org) >> receives a request from a user trying to call sip:u...@sip5060.net, with >> a sip: rather than sips: URI, should it (and will it) use the "SIPS+D2T" >> result, if no other result is available? >> >> Or would it ignore the NAPTR record and try to find the default SRV >> record such as _sip._udp.sip5060.net ? >> >> Should there be another NAPTR record to translate sip: to sips: using a >> regex perhaps, or would such a NAPTR be a bad thing? >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
