You’re using WebRTC. That requires ICE and DTLS-SRTP. You can’t get away from that. The media path is negotiated using ICE and candidates. If you’re behind a NAT, which you are in that environment, then rtp.conf has to be configured with ICE candidate mapping[1] or STUN has to be used to discover it - though using STUN then results in a reliance on an external STUN server.
I should also add this is effectively one of the most complex setups involving tons of different specs and technology.
[1] asterisk/configs/samples/rtp.conf.sample at master · asterisk/asterisk · GitHub