11 Jan
2013
11 Jan
'13
6:45 p.m.
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?
14 Jan
14 Jan
3:59 p.m.
The caller should use the NATPR and thus should use TLS. The SIPS+D2T
does not requires the URI to be a SIPS URI.
See also the thread
"NAPTR, SRV and sips vs. transport=tls" from 1.Dec.2012
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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6:23 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
>
8:38 p.m.
On 14.01.2013 18:23, Daniel Pocock wrote:
On 14/01/13 15:59, Klaus Darilion wrote:
I am not aware of such weird behavior. And if, there nothing you can do.
regards
Klaus
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? 
10:19 p.m.
14 jan 2013 kl. 18:23 skrev Daniel Pocock <daniel(a)pocock.com.au>au>:
On 14/01/13 15:59, Klaus Darilion wrote:
UPgrading from SIP: to SIPS has too many
implications so you should
propably never do that unless you really know what you want.
Transporting any SIP message over TLS doesn't require any changes
to the actual messages and work just fine. If I add that one single
NAPTR record I just state that if you want to SIP with my domain,
you always have to reach me over TLS. Period. So your proxy will
open a TLS session and that's fine and will not affect your first hop.
/O
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
>
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4 comments
3 participants
participants (3)
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Daniel Pocock -
Klaus Darilion -
Olle E. Johansson