Have a look at:
pjsip/src/pjsua-lib/pjsua_call.c: get_secure_level();

It seems to me that PJSIP does not support SIPS scheme in Record-Route. That
said, you have to put SIP and "transport=tls" when writing Record-Route.


Regards,

Brez

On Tue, Jul 5, 2011 at 12:16 PM, Brez Borland <brez...@gmail.com> wrote:

> Hi Inaki,
>
> On Tue, Jul 5, 2011 at 10:43 AM, Iñaki Baz Castillo <i...@aliax.net> wrote:
>
>> Hi, I'm experimenting a problem with a client (PJSIP) connecting to a
>> proxy via TLS:
>>
>> - The client uses "sip:" scheme in INVITE headers and "sip:" with
>> ";transport=tls" in Contact header. It is valid according to some RFC.
>>
>> - The proxy routes the INVITE via UDP and adds a Record-Route like this:
>>
>>    Record-Route: <sips:IP_PROXY:PORT;transport=tcp>
>>
>
> Did you try without the transport parameter at all, just "sips"?
>
>
> Regards,
>
> Brez
>
>
>>
>> - However the the client receives the 200 with the mirrored
>> Record-Route, it does not send the ACK via TLS (it just does not send
>> it at all).
>>
>> - But if the proxy would use a Record-Route as follows then the ACK is
>> correctly sent:
>>
>>    Record-Route: <sip:IP_PROXY:PORT;transport=tls>
>>
>>
>>
>> My question is: is it valid the first Record-Route my proxy is adding?
>> or should I use ";transport=tls" as in the working Record-Route just
>> because the INVITE has no "sips" scheme?
>>
>> Thanks a lot.
>>
>> --
>> Iñaki Baz Castillo
>> <i...@aliax.net>
>>
>> _______________________________________________
>> Sip-implementors mailing list
>> Sip-implementors@lists.cs.columbia.edu
>> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
>
>
>
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