WebRTC calls support for hold/unhold

Hi all,

Does Asterisk 11.6 + WebRTC module support hold/unhold of WebRTC call? I mean I use browser (Chrome 31) and latest JsSip (from hold/unhold branch) to instantiate the call but for some reason I can’t hold/unhold calls.
Developers of JsSip state:

  1. I have some doubts because sipml5.org allowed me to hold/unhold my call with all the same settings I use with JsSip.

I have this working with asterisk 11.6-cert, I’m using sipml5, no jssip, but, call hold/resume definitely works, albeit only over an unsecured websocket, still haven’t managed to get wss:// running, even with 12.2.0-rc1.