WebRTC Inbound Calls

I am struggling to solve an issue with audio working on inbound calls using WebRTC. Audio works in both directions for outbound (made via WebRTC). But when receiving a call, I cannot hear any audio from the caller.

Any ideas? I have tried changing configuration almost everywhere with no luck. Let me know what information you may need and I’d be happy to share!

Adding: I have narrowed it down to the packets are being received (as shown in chrome://webrtc-internals/) but they’re all being discarded…

Did you try using a stun server inside Asterisk SIP Settings > WebRTC Settings ?