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One Way Audio, Echo & Voice Quality Issues in VoIP – Complete Troubleshooting Guide

One-way audio, echo, robotic voices — these are the most frustrating VoIP problems. This comprehensive guide explains causes and step-by-step fixes.

Rahul Verma
Technical Support Lead
April 18, 20269 min read

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:

  1. Enable STUN in your softphone settings

STUN Server: stun.hexavoice.net:3478

  1. Enable "Use STUN for RTP" if available
  2. If behind corporate firewall, ask your IT team to allow RTP ports 10000-20000 (UDP)
  3. Try setting NAT Type = Symmetric or Full Cone in advanced settings

Echo Problems

Echo during calls usually comes from:

  1. Acoustic echo — your speaker audio is picked up by your microphone
  2. 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:

  1. Switch from WiFi to wired ethernet
  2. Enable QoS on your router — prioritize VoIP/SIP traffic
  3. Change codec from G.729 to G.711 if bandwidth allows
  4. Or switch to Opus codec — it handles packet loss much better

Complete Diagnostic Checklist

ProblemMost Likely CauseQuick Fix
One-way audioNAT/FirewallEnable STUN
No audio at allRTP ports blockedOpen UDP 10000-20000
EchoAcoustic / NetworkUse headset, enable AEC
Choppy voicePacket loss / JitterUse wired connection, QoS
Robotic voiceHigh jitterSwitch to Opus codec
Calls droppingRegistration timeoutEnable SIP keep-alives