One-way audio is a common bother businesses encounter when using VoIP phone systems. It occurs when one person can hear the other on a call, but the audio only travels in one direction.
To make a quick example, you might be chatting away to someone on the phone, and they might not even realise you’re even speaking! (The same can go both ways.)
Unlike traditional phone systems, VoIP relies on internet networks to transmit voice data. If there are configuration or network problems, voice packets may only travel in one direction, causing one-way audio.
This blog post will help you understand the most common causes and help identify and resolve the problem quickly.
What Is One-Way Audio in VoIP?
What happens in one-way audio is that audio packets do successfully travel in one direction, but they don’t make it back through the network path.
To send voice packets between devices, VoIP uses what’s called the Real-Time Transport Protocol (RTP). If the packets cannot return (because the return path is blocked or misconfigured), the receiving party cannot hear the audio.
So you’ll usually run into this issue in scenarios like:
- You can hear them, but they cannot hear you.
- They can hear you, but you cannot hear them.
- Calls connect, but audio only works one way.
Common Causes of VoIP One-Way Audio
Firewall or NAT Configuration Problems
Firewalls and routers are notorious for blocking the RTP ports that make your voice traffic work.
When Network Address (NAT) Translation is incorrectly configured, the VoIP system may send audio packets to the wrong IP address.
Common symptoms include:
- Calls connect, but audio only works one way
- External calls fail while internal calls work normally
- Issues occur only when calling outside the network
Problem: Firewall Blocking RTP Traffic
Solution: Allow VoIP Media Ports
As mentioned, VoIP uses RTP to transmit voice data, which, a lot of the time, is blocked by firewalls, so audio packets are only travelling in one direction.
Fix
Ensure the required RTP ports are open in the firewall and correctly forwarded to the VoIP system or PBX. Once the RTP stream can travel both ways, audio will return to normal.
Problem: SIP ALG Interference
Solution: Disable SIP ALG
A lot of routers have SIP Application Layer Gateway (ALG) turned on out of the box to help handle VoIP traffic. But this usually modifies SIP packets incorrectly.
This can lead to:
- One-way audio
- Call drops
- Calls failing to connect
Fix
We would highly recommend that you disable SIP ALG in the router or firewall settings.
Problem: NAT Configuration Issues
Solution: Configure NAT Traversal Correctly
Network Address Translation (NAT) allows multiple devices to share a single public IP address, but it can make things harder for VoIP traffic to travel
If NAT is misconfigured, the VoIP server may send audio packets to the wrong internal address. This means that if you’re chatting to someone, one or both of you might experience choppy audio (if at all) or, in some rare (but still possible) cases, in a multi-device network, a separate device might pick up the audio.
Fix
Configure NAT traversal properly using methods such as:
- STUN
- Static NAT rules
- Correct SIP server configuration
This ensures audio packets are routed back to the correct device.
Problem: Network Congestion or Packet Loss
Solution: Prioritise Voice Traffic
When you’ve got tons of devices on the internet at the same time, you’ll start to see lots of packet loss and jitter issues, affecting how voice data travels across the network.
Symptoms often include:
- Choppy audio
- Delayed voice
- Temporary one-way audio
Fix
Enable Quality of Service (QoS) on your router or network switch; this basically allows you to prioritise VoIP traffic over general data usage.
Problem: Incorrect Codec Configuration
Solution: Standardise Codec Settings
Codecs are used by VoIP systems to compress voice. Audio streams may not work right if the codecs are set up wrong or don’t work well together.
Fix
Use codecs that are widely used, like:
Keeping codec settings the same on all devices stops problems with audio transmission.
VoIP systems compress voice using codecs. If incompatible or inefficient codecs are configured, audio streams may fail to process correctly.
How Fuse 2 can help
At Fuse 2, we will work closely with you to identify your VoIP issues and resolve them quickly. With our team’s expertise and accreditations, we can ensure that networks are properly configured for voice traffic, from firewall rules and SIP settings to RTP port management and traffic prioritisation.
By proactively monitoring and optimising VoIP environments, we help our customers maintain clear, reliable communications without the disruption that issues like one-way audio can cause.
