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.

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.
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
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.
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.
Firewalls often allow SIP signaling but block RTP media ports. When this happens, calls connect but audio does not pass.
SIP ALG on routers or firewalls can modify SIP packets incorrectly, breaking audio streams.
If the PBX advertises the wrong IP address in SIP messages, remote endpoints will send audio to the wrong destination.
When the PBX is behind NAT, SIP and RTP ports must be forwarded correctly to the PBX.
Test:
Extension to extension call
Extension to external number
External number to extension
This determines whether the issue is internal, external, or both.
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.
Ensure the firewall allows UDP traffic for the RTP port range used by Grandstream.
If ports are blocked, audio will not pass.
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.
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.
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.
Test calls from an external network or mobile phone to confirm audio flow.
Monitoring packets or logs can help confirm RTP traffic.
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
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.
Assuming SIP connection guarantees audio
Opening SIP ports but forgetting RTP ports
Leaving SIP ALG enabled
Using incorrect public IP address in PBX settings
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.