Grandstream VoIP No Audio Issue – Troubleshooting Guide

Dec 25, 2025 15 mins read

Introduction

One of the most common and frustrating problems in VoIP systems is when calls connect successfully but there is no audio on one side or both sides. In Grandstream VoIP systems, this issue usually appears after deployment, firewall changes, NAT configuration, or SIP trunk setup.

This article explains the main reasons behind no-audio and one-way audio issues in Grandstream VoIP systems and provides a structured troubleshooting approach to fix them.

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Problem Description

Calls can be established successfully between extensions or with external numbers, but users cannot hear each other. In some cases, audio works in one direction only, while in others there is no audio at all.

This problem is rarely caused by the IP phones themselves. In most cases, it is related to network, NAT, firewall, or SIP/RTP configuration.

Common Symptoms

Calls connect normally but there is no audio
One-way audio only
Internal calls work but external calls have no audio
Audio works internally but not over SIP trunk
Audio cuts in and out during calls

Quick Checks Before Troubleshooting

First, test whether the issue occurs on internal calls or only external calls.
Check if the problem affects all phones or only specific extensions.
Confirm that audio devices (speaker and microphone) work on the phone.
Reboot the affected phone and PBX system.

These quick checks help isolate whether the issue is device-related or network-related.

Common Causes of No Audio Issues

NAT Configuration Problems

VoIP relies on RTP streams for audio. If NAT is not configured correctly, RTP packets may not reach the correct destination, causing no audio or one-way audio.

Firewall Blocking RTP Ports

Firewalls often allow SIP signaling but block RTP media ports. When this happens, calls connect but audio does not pass.

Incorrect SIP ALG Behavior

SIP ALG on routers or firewalls can modify SIP packets incorrectly, breaking audio streams.

Wrong Local Network or Public IP Settings

If the PBX advertises the wrong IP address in SIP messages, remote endpoints will send audio to the wrong destination.

Missing or Incorrect Port Forwarding

When the PBX is behind NAT, SIP and RTP ports must be forwarded correctly to the PBX.

Step-by-Step Troubleshooting

Step 1: Identify Where Audio Fails

Test:

  • Extension to extension call

  • Extension to external number

  • External number to extension

This determines whether the issue is internal, external, or both.

Step 2: Check RTP Port Range on Grandstream

Verify the RTP port range configured on the PBX.

Ensure:

  • RTP ports are defined correctly

  • The range matches firewall rules

Typical RTP range examples are 10000–20000.

Step 3: Allow RTP Ports in Firewall

Ensure the firewall allows UDP traffic for the RTP port range used by Grandstream.

If ports are blocked, audio will not pass.

Step 4: Check NAT Settings on PBX

Configure:

  • Correct local network subnet

  • Correct public IP or NAT IP

  • Enable NAT traversal options if required

Incorrect NAT settings are one of the top causes of no audio.

Step 5: Disable SIP ALG

If SIP ALG is enabled on the router or firewall, disable it and test again.

SIP ALG frequently causes one-way or no-audio problems.

Step 6: Verify Port Forwarding

If the PBX is behind NAT, forward:

  • SIP port (usually 5060 UDP)

  • RTP port range

Ensure forwarding is mapped to the correct internal IP address.

Step 7: Test from External Network

Test calls from an external network or mobile phone to confirm audio flow.

Monitoring packets or logs can help confirm RTP traffic.

Verification and Testing

After applying fixes:

  • Both parties should hear audio clearly

  • One-way audio issues should disappear

  • Internal and external calls should behave consistently

  • No RTP-related errors should appear in logs

Best Practices

Place the PBX on a stable network with minimal NAT complexity.
Use a dedicated VLAN for voice traffic.
Avoid double NAT whenever possible.
Keep firmware updated on PBX and IP phones.
Document SIP and RTP port usage.

Common Mistakes

Assuming SIP connection guarantees audio
Opening SIP ports but forgetting RTP ports
Leaving SIP ALG enabled
Using incorrect public IP address in PBX settings

Final Thoughts

No-audio issues in Grandstream VoIP systems are almost always related to network configuration rather than hardware failure. Understanding how SIP signaling and RTP media streams work together is key to resolving these problems. A structured troubleshooting approach ensures fast resolution and stable voice communication.

 

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