Basically i set up an asterisk server, connected to a sip provider to make calls to pstn or mobile networks. I have configured SIP to SIP properly because when i make calls from softphone e.g. Zoiper - Asterisk - Sip provider - Mobile network, the call is established and i can hear audio on both ends.
I want to use WebRTC so im using sipML5 as a client on localhost. I registered sip peer on sipml5 it works fine. I make a call to the softphone or to the PSTN/Mobile network and the call is established but no audio on both ends.
sipML5 gives me an error: onSetRemoteDescriptionError
DOMException: Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd.
I have enabled ice in rtp.conf and also in the peers in sip.conf. Also put google stun server in rtp.conf.
I can't figure out what the problem is. The problem is in WebRTC to SIP. I haven't installed webrtc2sip gateway by doubango. i am not sure if i should install it since im using asterisk 13.
Any idea what might the problem be?
So, just posting the answer to this for anyone that might need it in the future. Basically i was working on localhost without https, for WebRTC is mandatory having https, and in order to make the calls i enabled rtc breaker which lets you make calls even if it is not https connection. So after i just made the connection https, disabled rtc breaker and now everything works as expected. Audio is produced imediately after picking up the phone.