| Cause | Explanation | |-------|-------------| | | Username, password, or domain typo | | Network restrictions | Firewall/proxy blocking WebRTC or SIP traffic (usually UDP/TCP 5060, or ports 10000–20000 for RTP) | | Server unreachable | Wrong server address, or server down | | Expired registration | Line registered earlier but timer ran out due to network change or long inactivity | | Multiple registrations | Same SIP account registered on another device (some servers kick the older one) | | WebRTC-specific issue | Browser lacks permissions for microphone/camera, or SSL certificate problem (WebRTC requires HTTPS or localhost) |
| Platform | Common Fix | |----------|-------------| | | Go to Settings → SIP Account → Re-register. Ensure extension is allowed for WebRTC in 3CX console. | | FusionPBX / FreeSWITCH | Check sip_profile → ws-binding and wss-binding are enabled. | | Asterisk (PJSIP) | Ensure websocket_enabled = yes and transport=wss in pjsip.conf. | | Chrome / Edge | Disable any VPN/proxy extensions. Try Incognito mode. | | Firefox | Check about:config → media.peerconnection.enabled is true . | webrtc line is not registered
Verify these in your WebRTC client settings: | Cause | Explanation | |-------|-------------| | |
// Register the WebRTC line pc.onaddstream = (event) => console.log("WebRTC line registered"); ; | | Asterisk (PJSIP) | Ensure websocket_enabled =
The most frequent cause of registration failure is . Modern office networks are often protected by deep-packet inspection firewalls or SIP Application Layer Gateways (ALGs). While intended to be helpful, SIP ALGs often corrupt the headers of WebRTC packets, leading the server to reject the registration attempt. Similarly, strict firewall rules may block the specific ports (usually 5060 for SIP or various UDP ports for media) required for the connection to stay alive.
// Add a stream to the peer connection pc.addStream(stream);