Quick Reference

Error Audio One Way
Severity Critical
Component Network

Audio One Way

What Causes This Error

The call is connected but audio only flows in one direction — either the agent can’t hear the caller or the caller can’t hear the agent. Root cause is almost always NAT traversal issues with RTP media streams. The SIP signaling succeeded but RTP media packets are being blocked or misrouted by NAT or firewall rules.

How to Fix It

  • 1 Check Asterisk RTP configuration: cat /etc/asterisk/rtp.conf

  • 2 Verify RTP port range is open in firewall: iptables -L | grep 10000 (or whatever range is configured)

  • 3 Enable NAT option in sip.conf for affected peers: nat=force_rport,comedia

  • 4 Check if STUN/TURN is configured for external agents

  • 5 Test with canreinvite=no in sip.conf to force media through Asterisk

  • 6 Verify externip setting in sip.conf matches your actual external IP

  • 7 Capture RTP traffic: tcpdump -i eth0 -nn udp port 10000:20000

  • 8 Check if the problem is carrier-side or agent-side by testing both audio directions

How to Prevent It

Configure NAT traversal properly during initial setup. Use a VPN for remote agents to eliminate NAT issues. Keep RTP port range firewall rules documented. Test audio quality after any network or firewall changes.

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Frequently Asked Questions

What causes the VICIdial "Audio One Way" error? +

The call is connected but audio only flows in one direction — either the agent can't hear the caller or the caller can't hear the agent. Root cause is almost always NAT traversal issues with RTP media streams. The SIP signaling succeeded but RTP media packets are being blocked or misrouted by NAT or firewall rules.

How do I fix the VICIdial "Audio One Way" error? +

Check Asterisk RTP configuration: cat /etc/asterisk/rtp.conf Then: Verify RTP port range is open in firewall: iptables -L | grep 10000 (or whatever range is configured) Then: Enable NAT option in sip.conf for affected peers: nat=force_rport,comedia Then: Check if STUN/TURN is configured for external agents Then: Test with canreinvite=no in sip.conf to force media through Asterisk Then: Verify externip setting in sip.conf matches your actual external IP Then: Capture RTP traffic: tcpdump -i eth0 -nn udp port 10000:20000 Then: Check if the problem is carrier-side or agent-side by testing both audio directions

How can I prevent the "Audio One Way" error in VICIdial? +

Configure NAT traversal properly during initial setup. Use a VPN for remote agents to eliminate NAT issues. Keep RTP port range firewall rules documented. Test audio quality after any network or firewall changes.

Related Errors

WebRTC Connection Failed View error → SIP Registration Failed View error → Carrier Trunk Full View error →

Part of the Complete VICIdial Implementation Guide

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