Are you using the media gateway webrtc2sip or connecting directly to asterisk? Also share the complete sip debug and the JavaScript log from your browser.
==session event = m_stream_audio_remote_added
Uncaught TypeError: Cannot set property 'src' of null call.js:564 To be honest i don’t know if that is part of your issue but comparing with my workings js log I don’t have those.
Are you using your own code or using the sipml5 example page?
As far I know ICE servers are needed by SIPML5, latest chrome version works and you need to debug deeper, with js and asterisk side like network properties, Nat & RTP debug. Finally test the sipml5 original demo, if it works then you need to take a look at your code.