SRTP Support in ASTPP

I need to add SRTP support but I can’t find default.xml file in ASTPP.
According to freeswitch I should edit conf/directory/default.xml and change the dial-string param to:

please advise.

Any sip profile related modification you can do from Switch > Sip Profile. ASTPP is building the profile configuration XML from database and then push it into FS.

Thank you for your support.
I did the required modification and now when calling peer tp peer the calls are not connected.
the freeswitch log is:
mod_dptools.c:3653 Originate Failed. Cause: SERVICE_NOT_IMPLEMENTED

Can you advise please!

Both the end points should support and offer encrypted media.

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