One Way Audio — Most Common Cause: NAT
One-way audio (you can hear them but they can't hear you, or vice versa) is almost always a NAT traversal problem.
What is NAT?
When you're behind a router, your device has a private IP (192.168.x.x). The SIP server sees your public IP. If the RTP audio packets are sent to the wrong IP address, you get one-way audio.
Fix for One Way Audio:
- Enable STUN in your softphone settings
STUN Server: stun.hexavoice.net:3478
- Enable "Use STUN for RTP" if available
- If behind corporate firewall, ask your IT team to allow RTP ports 10000-20000 (UDP)
- Try setting NAT Type = Symmetric or Full Cone in advanced settings
Echo Problems
Echo during calls usually comes from:
- Acoustic echo — your speaker audio is picked up by your microphone
- Network echo — delay in the network causes echo
Fixing Acoustic Echo:
- Use a headset instead of speakers
- Enable Echo Cancellation (AEC) in your softphone
- Reduce microphone volume
Fixing Network Echo:
- Check your network latency (ping to sip.hexavoice.net should be under 150ms)
- Use a wired connection instead of WiFi
- Upgrade your internet if jitter is over 30ms
Robotic / Choppy Voice
Caused by high packet loss or jitter.
Diagnosis:
Run a VoIP test: https://test.hexavoice.net
Fixes:
- Switch from WiFi to wired ethernet
- Enable QoS on your router — prioritize VoIP/SIP traffic
- Change codec from G.729 to G.711 if bandwidth allows
- Or switch to Opus codec — it handles packet loss much better
Complete Diagnostic Checklist
| Problem | Most Likely Cause | Quick Fix |
|---|---|---|
| One-way audio | NAT/Firewall | Enable STUN |
| No audio at all | RTP ports blocked | Open UDP 10000-20000 |
| Echo | Acoustic / Network | Use headset, enable AEC |
| Choppy voice | Packet loss / Jitter | Use wired connection, QoS |
| Robotic voice | High jitter | Switch to Opus codec |
| Calls dropping | Registration timeout | Enable SIP keep-alives |