A solution requires understanding what is going on. First start by simplifying your scenario. Does calling from the webphone to Asterisk that just does a Playback work? If not, you’ve now narrowed it down. What is the SIP trace? What is the network arrangement? Is ICE negotiation and DTLS negotiation completing? Are RTP packets being sent from Asterisk to it?
WebRTC is built on various specifications and technology, and if you’re using/deploying it then most likely it will fail to work in some cases. An understanding of these is required to diagnose and understand what is going on.