@RobMat try debugging the rtp flow using rtp set debug on to observe which if it’s not sending to private ip instead. You might have to mess around with your client application too. Add stun address to the app’s settings. Try testing with different clients, like AntiSip. Zoiper and Linphone didn’t work for me when I was trying this out. Our usecase involved video as well though.
Not sure if it will help but still here’s our issue thread Webrtc SipML 5 audio issue + asterisk