14 jan 2013 kl. 18:23 skrev Daniel Pocock :
> 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 pr
On 14.01.2013 18:23, Daniel Pocock wrote:
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 w
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 an
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
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. INNAPTR10 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 Kamaili
