Good call quality starts with a solid network setup. dialnote uses WebRTC technology for voice calls, which means your network needs to allow certain connections to pass through. Most networks work out of the box, but corporate firewalls and strict security settings sometimes need adjustments.
Network Requirements#
dialnote's voice calls require the following:
Bandwidth:
- Minimum: 100 Kbps upload and download per concurrent call
- Recommended: 250 Kbps or higher for best quality
- For multiple simultaneous calls, multiply by the number of active calls
Ports and protocols:
- HTTPS (port 443): API communication and signaling
- UDP ports 10000-60000: Voice media traffic (RTP/SRTP)
- TCP port 443: Fallback when UDP is blocked (TURN over TLS)
Domains to allow:
*.dialnote.com- Application services- Our voice provider's media and signaling domains (contact support for the current list)
Quick test
Run the Audio Test from Settings → Your Account → Audio Test to check if your network can reach the voice servers. The network check sends a quick request to the voice servers and measures the round-trip latency. See Call Quality for device-level fixes.
Firewall Configuration#
If you're behind a corporate firewall, you may need to whitelist specific IP ranges. WebRTC uses STUN and TURN servers to establish connections, especially when devices are behind NAT (Network Address Translation).
What to allow:
- Outbound UDP traffic on ports 10000-60000
- Outbound TCP traffic on port 443 (for TURN fallback)
- The voice servers' IP ranges (contact support for the current list)
Signs your firewall is blocking calls:
- Calls connect but no audio in one or both directions
- Calls drop after a few seconds
- The Network Connection check in the Audio Test times out ("Network timeout") or shows a slow result. A pass doesn't rule out a firewall.
- High latency warnings (over 500ms)
VPN considerations
Some VPNs route all traffic through their servers, which can add latency and cause audio issues. If you're experiencing problems, try disconnecting from VPN or configuring split tunneling to exclude dialnote traffic.
Optimizing Wi-Fi Performance#
Wi-Fi networks can introduce latency and packet loss that affects call quality. Here's how to get the best performance:
Router placement:
- Position your router centrally, away from walls and metal objects
- Avoid placing it near microwaves, cordless phones, or baby monitors
- Elevate the router if possible—floor placement reduces signal quality
Channel selection:
- Use 5GHz bands when available (less interference than 2.4GHz)
- Check for channel congestion using a Wi-Fi analyzer app
- Set your router to auto-select channels or manually pick the least congested
Quality of Service (QoS): If your router supports QoS, prioritize voice traffic. Look for settings labeled "VoIP priority" or "Real-time traffic" and enable them for devices running dialnote.
Testing Your Connection#
dialnote includes built-in diagnostics to verify your setup works correctly.
Running the Audio Test:
- Go to Settings → Your Account → Audio Test
- Click Start Test (it shows Run Again after the first run)
- The system checks browser compatibility, microphone, speaker, and network connectivity in order
- Review results—all four checks should pass
The browser check confirms your browser supports the building blocks for WebRTC calls: RTCPeerConnection, microphone access, and the Web Audio API. If any of these are missing, switch to an up-to-date Chrome, Firefox, Safari, or Edge. If the browser check fails, the other three checks show Skipped.
Understanding network test results:
- Excellent (under 200ms latency): Best call quality
- Good (200-500ms): Acceptable quality for most calls
- Slow (over 500ms): May experience delays and echo. The check still shows Passed, so read the latency message, not just the badge.
- Failed/Timeout: Check firewall settings and internet connection
Browser requirements
dialnote requires WebRTC support. Use Chrome, Firefox, Safari, or Edge. Older browsers or heavily restricted corporate browsers may not work.
Common Network Issues#
One-way audio (you can hear them, they can't hear you):
- Check microphone permissions in browser settings
- Verify your firewall allows UDP traffic
- Try a different network to isolate the issue
Choppy or robotic audio:
- Reduce bandwidth usage by other applications
- Switch from Wi-Fi to wired Ethernet if possible
- Check for network congestion during peak hours
Calls dropping unexpectedly:
- Test your internet stability (try a speed test that measures consistency)
- Look for intermittent connection drops in your router logs
- Contact your ISP if problems persist
High echo or feedback:
- Use headphones instead of speakers
- Reduce speaker volume
- Enable echo cancellation in your operating system's audio settings
What's Next#
- Call Quality — pick the right microphone and speaker, and fix choppy or one-way audio
- Making Calls — in-call controls and the dialer
- Mobile App — call quality tips for iOS and Android
- Contact Support — reach the team if calls still won't connect