[SR-Users] NAPTR for TLS only

[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 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 at 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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