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SRTP output came out, but crypto suite was not discussed with kamailio rtpengine

I am trying to integrate webrtc->kamailio->asterisk to call from web browser.

I am using kamailio configuration file from caruizdiaz and a chrome browser with sipml5 and an asterisk as a media server. So far, I received the pstn number call via sip trunking from the browser, but there is no sound, and the following error message appears in the rtpengine log. "SRTP output came out, but not a single crypto packet was negotiated"

I think this is an error when kamailio cannot establish DTLS negotiation and audio packets are discarded.

My question is how to make DTLS negotiations successful, be it a chrome or asterisk error? because I use the RTP / AVP profile to send multimedia packets to an asterisk.

I have included my log here kamailio-webrtc-log

THANKS WITHIN THE FRAMEWORK.

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