14 Jan
2013
14 Jan
'13
12:23 p.m.
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?
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>