I am wondering if it is possible to stream a conference call to a public facing website? Is there some easy “freepbx” way of doing this? Can I just access the webrtc stream by authenticating to it somehow? I have attempted to use sipML5 to connect but I am getting an authentication error. This is after I have added noload = chan_sip.so to /etc/asterisk/modules.conf.
I now also see that the modules.conf file has …edited itself back.
realistically you will need to use wss with a real certificate., then your client can connect on wss:// port 8089 by default with correct credentials. but Asterisk by itself doesn’t provide any webrtc conferencing, perhaps
will give some backgroind in which direction to go
That’s because as stated neither chan_sip nor chan_pjsip can directly negotiate wss: sip is a different protocol. Also as the link explains asterisk listens for wss: on port 8089 so you will need to use that also when using sipMl5, But feel free to knock yourself out trying
Transport=wss only , you will also need dtls enabled and as noted a working public certificate, but its still only an endpoint and can’t host a conference