Without Stun/Turn servers configured, the only candidates you might get from onicecandidate are "host" type, which technically is your local ip address.
For my case :
type:host # address:192.168.1.56 # port:57934
type:host # address:192.168.1.56 # port:9
I have a WebRTC UserAgent (SIPJS Library) behind a firewall, connected to a B2BUA (Freepbx to be more precise) via 8089 wss for signaling. The asterisk server is cloud based, with a public ip address (No stun).
I have not configured any Stun/Turn Server on my SIPJs peerConnectionConfiguration, and still my webRTC client is able to share RTP media with Asterisk with no problem.
How is that possible ? is there any built-in stun server on the browser in case no stun server is configured ? Do i really need a Stun server in such use case ?
Thank you
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Thank you
In other words if my Asterisk is on public IP , i dont need to use any Stun/Turn servers on my webRTC clients ? is it safe to disable it ?
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