(Running Asterisk 16.0.0)
When using WebRTC and Dial/ConfBridge with PJSIP if you end the call with a normal BYE all channels clear up as expected. However if you refresh the web page the channels are left open
You will get output such as:
Removed contact 'sip:fdea6gog@XXX.XX.XXX.XXX:13069;transport=ws' from AOR 'foo' due to transport shutdown Contact foo/sip:fdea6gog@XXX.XX.XXX.XXX:13069;transport=ws has been deleted == Endpoint foo is now Unreachable == WebSocket connection from 'XXX.XX.XXX.XXX:13069' closed
WARNING: res_http_websocket.c:516 ws_safe_read: Web socket closed abruptly
The output makes sense and is expected, but shouldn’t it clear up all the channels too?
It is trickier to reproduce with Dial because both parties need to refreshing at the same time; also have yet to reproduce on Chrome, just FireFox