WebRTC with Asterisk 11?

Tonight I have tried two WebRTC clients (JsSIP and sipML5) with Asterisk 11 and get them working – ulaw codec and echo test works, but with one-way audio if I call SIP softphone.

I get WebRTC clients working – it’s docummented how in following Asterisk wiki articles: Asterisk WebRTC Support and WebRTC tutorial using SIPML5.

Only Chrome works well, Firefox won’t.