Tonight I have tried two WebRTC clients (JsSIP and sipML5) with Asterisk 11 and get both of them working — echo test calls with ulaw (g711u) codec works, but with one-way audio if I call from WebRTC to the SIP softphone.
I’ve followed Asterisk wiki articles: Asterisk WebRTC Support and WebRTC tutorial using SIPML5 to configure WebRTC.
Good news!