WebRTC with Asterisk 11?

Tonight, I tried two WebRTC clients (JsSIP and sipML5) with Asterisk 11. I was able to get both of them working for echo test calls with the ulaw (g711u) codec. However, when I called from WebRTC to the SIP softphone, there was only one-way audio.

To configure WebRTC, I followed the Asterisk wiki articles: Asterisk WebRTC Support and WebRTC tutorial using SIPML5.