I am trying to integrate webrtc->kamailio->asterisk to call from web browser.
I am using kamailio configuration file from caruizdiaz and chrome browser with sipml5 and asterisk as media server. Till now I have achieved to call to pstn numbers through sip trunking from browser but there is no audio and in the log rtpengine is showing the following error message. "SRTP output wanted, but no crypto suite was negotiated"
I think it is the error where kamailio is not able to establish DTLS negotiation and audio packets are dropped.
My question is how to make DTLS negotiation successful, whether it is error from chrome side or asterisk? because I am using RTP/AVP profile to send media packets to asterisk.
I have included my log here kamailio-webrtc-log
THANKS IN ADVANCE.