SIP call issues in webRTC configured asterisk server (version 13.30_

Hello guys,

I need to enable webRTC configurations in asterisk server, in such a way that I can able to make voice calls from any browser.

I have setup asterisk server in aws instance (centos-7, asterisk-13.30).

I have configured all the necessary configurations using guide from asterisk official site.

I have successfully able to make voice calls, but while trying to attend the call, I am facing the following issue.

"process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio".

Can anyone please help to resolve this one.


Please post full SIP debug from Asterisk.