Why Webphone (formerly Mizu Webphone) Is the Best Browser SIP Client for Teams

Webphone (formerly Mizu Webphone): Top Tips for Troubleshooting and Optimization

1. Quick overview

Webphone is a browser-based SIP/WebRTC softphone that runs in modern browsers. It lets users make and receive VoIP calls without installing native apps, relying on SIP signaling and media capabilities in the browser.

2. Common problems and quick fixes

• No audio / one-way audio:

  • Check microphone/speaker permissions in the browser and OS.
  • Verify codec support (prefer OPUS for WebRTC).
  • Ensure NAT traversal (STUN/TURN) is configured; add a TURN server if media fails to flow.
  • Confirm ports (RTP range) are not being blocked by firewall.

• Cannot register / authentication errors:

  • Confirm SIP username, password, and domain are correct.
  • Check that SIP over WebSocket (WSS) endpoint URL is correct and uses valid TLS certificate.
  • Inspect server logs for authentication failures and browser console for detailed errors.

• Call drops / poor call quality:

  • Test network latency, jitter, and packet loss. Prioritize RTP traffic (QoS) on the network.
  • Use OPUS adaptive bitrate and ensure bandwidth is sufficient (at least 30–100 kbps per call).
  • Reduce video resolution or disable video to save bandwidth.

• Browser compatibility issues:

  • Use Chromium-based browsers or Firefox — ensure browser is up to date.
  • Confirm WebRTC and WebSocket support; older browsers may lack necessary APIs.

3. Diagnostic steps and tools

• Browser console & network tab: Look for SIP errors, WSS connection status, and failed resource requests.
• WebRTC internals (chrome://webrtc-internals): Examine ICE candidates, connection state, stats for packet loss, RTT, and codecs.
• SIP traces (server): Compare SIP INVITE/200 OK/ACK flows to identify signaling problems.
• Packet capture (Wireshark): Inspect RTP/RTCP and SIP over WS to debug media and signaling paths.
• Online speed tests and ping/traceroute: Check internet quality and route to SIP/ TURN servers.

4. Configuration tips for reliability

• Use WSS and valid TLS certificates: Prevent mixed-content issues and ensure secure signaling.
• Enable TURN servers: Provide fallback media relay when direct peer-to-peer fails. Use authenticated TURN.
• Optimize ICE candidate gathering: Prefer host and relay candidates; configure proper STUN/TURN priorities.
• Set appropriate RTP port ranges: Match firewall/NAT rules and document the range for network admins.
• Session timers and keepalives: Configure SIP session timers and periodic keepalive (e.g., OPTIONS or CRLF) to maintain NAT bindings.

5. Performance tuning

• Codec selection: Prefer OPUS for audio; use VP8/VP9 or H.264 for video depending on client support and CPU.
• Adaptive bitrate & silence suppression: Enable features that reduce bandwidth during silence and adapt to network changes.
• Limit concurrent calls per user: Prevent resource exhaustion on browsers and servers.
• Hardware acceleration: Encourage use of devices/browsers that support hardware encoding for video.

6. Security best practices

• Use strong authentication and rate-limiting to prevent SIP attacks.
• Keep server and signaling software updated.
• Enforce TLS/WSS and SRTP where possible.
• Monitor logs for unusual registration or call patterns.

7. User-facing troubleshooting checklist

1. Confirm microphone and speaker devices are selected and permitted in browser.
2. Reload the page and retry; clear cache if needed.
3. Try a different browser or private/incognito window.
4. Restart the device or network equipment.
5. If issues persist, capture console logs and provide them to your VoIP admin with timestamps.

8. When to escalate to your VoIP admin

• Repeated registration failures across multiple users.
• Server-side certificate errors or WSS connection failures.
• Consistent poor MOS scores, packet loss, or network-level issues.
• Authentication failures indicating possible account or configuration problems.

9. Summary

Following these troubleshooting steps and optimizations—checking permissions, using WSS/TURN, monitoring WebRTC internals, and tuning codecs and network settings—will resolve most Webphone issues and improve call quality.