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:user@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?
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