WebRTC browser calls FreePBX 13 (Stable)

Unaware that WSS is not supposed to work, I got it working today on FreePBX 13 and Asterisk 11.13 through no special effort other than to enable Asterisk’s HTTPS server on 8089 and supply a valid certificate. I was already using HTTPS in Apache for admin and UCP. What is it that makes this unsupported–or was it fixed in the last few weeks?

I can make outgoing calls but cannot receive calls. The WebRTC client immediately appears unregistered to Asterisk by failing to respond to a qualify:

[2016-01-27 20:57:32] NOTICE[12903]: chan_sip.c:29479 sip_poke_noanswer: Peer '991101' is now UNREACHABLE! Last qualify: 0

I can’t tell whether this is a WebRTC issue or a typical network (NAT, firewall) issue, though my SIP phones on the same network are experiencing no difficulties with registration.