I’m following this tutorial (
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5) 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.

*;sip.conf:*
[general]
udpbindaddr=0.0.0.0:5060
realm.201.0.106 ;replace with your Asterisk server public IP address or host transport=udp,ws,wss tlsenable=yes tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt tlscipher=ALL
tlsclientmethod=tlsv1

[6000]
host=dynamic secret=mysecret context