No Audio On SIP Over WebRTC

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I’m following this tutorial ( to deploy WebRTC support but I’m having an issue with RTP when the WebRTC
softphone is behind NAT.

In my scenario, the Asterisk server is running a public IPv4, and the softphone is behind NAT. I can register and make a call normally, but I
don’t get any audio in neither way (Asterisk/softphone and softphone/Asterisk). Using the very same config files but having the softphone and Asterisk on the same network it works fine.

Any tips on how to solve this? Here’s my relevant files.

realm.201.0.106 ;replace with your Asterisk server public IP address or host transport=udp,ws,wss tlsenable=yes tlsbindaddr=
tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt tlscipher=ALL

host=dynamic secret=mysecret context