FreePBX webrtc STUN issue when calling through VPN

I am using FreePBX version 16 the sip and extension are all configured in pjsip settings. The issue seems to be with NAT STUN when calling in public network. The call gets connected , no voice is transferred and gets disconnected after 20 to 30 seconds.

When making webrtc call through UCP from mobile to another mobile using each of their mobile data from their respective service provider then the calls are working fine.

Issue occurs when making calls through VPN or through wifi.

Can someone help me if it is some issue with my asterisk sip settings i have attached screenshots of the same.

You need to have your VPN subnet and wifi subnets (if you have devices making calls on the wifi subnet) listed in the Local Networks section.

Not sure that the entry is doing you any favors either.

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.