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NAT
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NAT issues
When you use NAT you may face some audio issues with sangoma phone desktop clients, such as "one way audio" or "Calls dropping after XX Seconds". This issue can be resolved with below mentioned settings.
Adding STUN server address
Browse to Settings → Astersik Asterisk SIP settings page and add STUN Server Address under WebRTC Settings. You can find the list of stun servers at STUN Servers.
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Adding ICE host candidates
Browse to Settings → Astersik Asterisk SIP settings page and add Candidates under ICE Host Candidates. Local address will be your server's local LAN IP address and advertised address will be your public address.
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