On Tue, Aug 18, 2015 at 3:12 AM, Chirag Desai djchill...@gmail.com wrote:
Hi all,
I'm using Kamailio + Asterisk 13 (PJSIP), where Kamailio (using rtpengine)
acts as the registrar and forwards all calls to Asterisk.
This works fine when using udp / tcp and RTP. When switching to TLS/SRTP,
Hi all,
I'm using Kamailio + Asterisk 13 (PJSIP), where Kamailio (using rtpengine)
acts as the registrar and forwards all calls to Asterisk.
This works fine when using udp / tcp and RTP. When switching to TLS/SRTP,
the call is set up correctly, however, I get no audio.
When I skip kamailio and