Troubleshooting

Troubleshooting Audio Quality Issues

Updated 1 week ago
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Introduction

Audio quality issues can significantly impact business communications and customer satisfaction. This comprehensive guide will help you identify, diagnose, and resolve common audio problems in your VoIP system. Whether you're experiencing echo, choppy audio, static, or complete audio loss, this guide provides step-by-step solutions.

Quick Tip

Before diving into complex troubleshooting, always start with the basics: check your internet connection, restart your device, and verify your audio settings. These simple steps resolve 60% of audio issues.

Common Audio Issues Overview

Understanding the different types of audio problems helps you diagnose issues faster:

Echo or Feedback

High
Common causes:
  • Speaker volume too high
  • Microphone sensitivity
  • Acoustic environment

Choppy/Robotic Audio

High
Common causes:
  • Packet loss
  • Jitter
  • Bandwidth congestion

Static/Crackling

Medium
Common causes:
  • Electrical interference
  • Faulty hardware
  • Poor connections

One-Way Audio

Critical
Common causes:
  • Firewall blocking
  • NAT issues
  • SIP ALG enabled

Low Volume

Low
Common causes:
  • Device settings
  • Codec mismatch
  • Gain settings

Echo and Feedback

Echo occurs when you hear your own voice repeated during a call. This is one of the most common and disruptive audio issues.

1

Check Speaker Volume

High speaker volume can cause sound to leak back into the microphone.

Solution:

  • Lower speaker volume to 50-70%
  • Use headphones instead of speakers
  • Increase physical distance between speaker and mic
2

Enable Echo Cancellation

Most VoIP systems have built-in echo cancellation features.

Settings → Audio → Advanced → Enable Echo Cancellation

Also enable Noise Suppression and Automatic Gain Control if available.

3

Optimize Room Acoustics

Hard surfaces cause sound reflections that can create echo. Add soft materials (carpets, curtains, acoustic panels) to reduce echo in conference rooms.

Choppy or Robotic Audio

Choppy, robotic, or fragmented audio is typically caused by network issues such as packet loss, jitter, or insufficient bandwidth.

Network Quality Requirements for VoIP

Bandwidth
100 kbps
Per concurrent call
Packet Loss
< 1%
Maximum acceptable
Jitter
< 30ms
Maximum acceptable

Run Network Diagnostics

Use built-in diagnostic tools to check your network quality:

Admin Portal → Diagnostics → Network Test → Run VoIP Quality Check

Enable Quality of Service (QoS)

QoS prioritizes VoIP traffic over other internet activities:

  1. Log into your router admin panel
  2. Navigate to QoS settings
  3. Create a rule prioritizing VoIP ports (UDP 5060-5061, 10000-20000)
  4. Set VoIP traffic to highest priority

Optimize Bandwidth Usage

  • Close unnecessary applications consuming bandwidth
  • Pause large downloads/uploads during calls
  • Use wired connection instead of WiFi when possible
  • Upgrade internet plan if consistently insufficient

Static and Noise

Static, crackling, or background noise can be caused by electrical interference, hardware problems, or environmental factors.

Hardware Issues

  • Test with different headset/microphone
  • Check for loose cable connections
  • Try different USB ports
  • Replace damaged cables

Electrical Interference

  • Move away from electrical devices
  • Use shielded cables
  • Check for fluorescent lighting nearby
  • Avoid power strip overloading

Microphone Settings

  • Reduce microphone gain/sensitivity
  • Enable noise suppression
  • Check for correct input device selection
  • Update audio drivers

Codec Configuration

  • Switch to higher quality codec (G.722)
  • Disable low-quality codecs (G.729)
  • Ensure codec compatibility
  • Test different codec settings

One-Way Audio

One-way audio means one party can hear the other, but not vice versa. This is typically a firewall or NAT configuration issue.

Critical: Firewall Configuration

One-way audio is almost always caused by firewall or router settings blocking RTP (media) traffic.

Required Firewall Ports:

  • SIP Signaling: UDP 5060-5061
  • RTP Media: UDP 10000-20000 (or your configured range)
  • STUN: UDP 3478

1. Disable SIP ALG

SIP Application Layer Gateway (ALG) often causes more problems than it solves:

Router Settings → Advanced → SIP ALG → Disable

2. Configure Port Forwarding

Forward VoIP traffic to your PBX/device:

ServiceProtocolPort Range
SIPUDP5060-5061
RTPUDP10000-20000

3. Check NAT Settings

Ensure your VoIP device is configured for your NAT environment:

  • Enable STUN (Session Traversal Utilities for NAT)
  • Configure external IP address if using static IP
  • Enable NAT keepalive packets

Low Volume Issues

If call volume is consistently low, check these settings:

System Level

  • Check OS master volume (should be 80-100%)
  • Verify correct audio output device selected
  • Check communication device volume separately
  • Update audio drivers to latest version

Application Level

  • Adjust VoIP app volume settings
  • Increase microphone gain (not too high)
  • Enable automatic gain control (AGC)
  • Test with different audio devices

Network Level

  • Check if codec supports adequate bitrate
  • Verify RTP packet size settings
  • Ensure sufficient bandwidth available
  • Check for packet loss reducing audio quality

Hardware Level

  • Test with different headset/handset
  • Check for hardware volume controls
  • Verify proper cable connections
  • Try different USB/audio ports

Network Diagnostics Tools

Use these tools to diagnose network-related audio issues:

Built-in VoIP Testing

Most VoIP systems include diagnostic tools:

Admin Portal → Tools → Network Analyzer → Run Test

This will test: latency, jitter, packet loss, bandwidth, and MOS score.

Understanding MOS Score

4.0-5.0
Excellent
3.5-4.0
Good
3.0-3.5
Fair
2.5-3.0
Poor
<2.5
Bad

MOS (Mean Opinion Score) measures call quality on a scale of 1-5. Aim for 4.0 or higher.

Command Line Tools

Test Latency (Ping):

ping -n 50 [your-voip-server]

Trace Network Path:

tracert [your-voip-server]

Advanced Solutions

Dedicated VoIP VLAN

Separate VoIP traffic on its own VLAN for improved quality and security. Requires managed switches.

Session Border Controller (SBC)

Enterprise solution that manages VoIP traffic at network perimeter, handling NAT traversal, security, and quality monitoring.

Implement MPLS

Multi-Protocol Label Switching provides guaranteed bandwidth and low latency for VoIP traffic across WAN connections.

Redundant Internet Connections

Use multiple ISPs with automatic failover to ensure call quality even if primary connection degrades.

Prevention Tips

Proactive Measures

  • Regularly test call quality
  • Monitor network performance metrics
  • Keep firmware and software updated
  • Perform quarterly network audits
  • Document baseline performance metrics
  • Train users on best practices

Maintenance Schedule

  • Weekly: Review call quality reports
  • Monthly: Update device firmware
  • Quarterly: Network infrastructure audit
  • Bi-annually: Hardware inspection
  • Annually: Complete system review
  • As-needed: User training refreshers

Still Experiencing Audio Issues?

Our technical support team is available 24/7 to help diagnose and resolve complex audio quality problems.

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