Webrtc stream access

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.

webrtc doesnt use chan_sip nor chan_pjsip


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

I’m not sure that is accurate because when I attempt to connect using sipML5, I see this on the asterisk server:

[2020-04-05 11:46:41] NOTICE[8268]: chan_sip.c:29029 handle_request_register: Registration from ‘"601"sip:[email protected]’ failed for xxxxxxxxx:58872’ - Wrong password

Not sure where the password is for this.

If I disable chan_sip, it will use pjsip, but then give me an authentication error.

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 :wink:

I am using wss on port 8089. I never said I wasn’t.

If chan_sip or chan_pjsip has nothing to do with, why does asterisk kick back a chan_sip wrong password error when I try to connect?

Like I asked before, where is that password? Or why is chan_sip saying bad password?

Probably because you have told the 601 extension to use a protocol other than wss://

Where would one make that change? Because I havn’t changed much of anything.

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

SIP over web socket is still SIP.

True, after it gets through the transport layer.

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