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Overview
The VoIP Network Performance Tester is a comprehensive tool designed to evaluate your network's capability to handle Voice over IP (VoIP) traffic. It simulates multiple concurrent users making VoIP calls and measures key performance metrics that directly impact call quality.
Why Test VoIP Performance?
VoIP calls are sensitive to network conditions. Poor network performance can result in:
- Choppy or robotic-sounding audio
- Delays in conversation (high latency)
- Dropped calls
- Echo or audio artifacts
How to Use the Tester
1. Configure Test Parameters
- Endpoint: Set to 8.8.8.8 (Google DNS) by default for reliable connectivity testing
- Codec: Choose the audio codec your VoIP system uses
- Test Duration: How long to run the test (10-300 seconds)
- Concurrent Users: Number of simultaneous VoIP calls to simulate (1-200)
- Simulation Mode: How users are distributed during the test
2. Choose Simulation Mode
- Steady Load: Maintains constant user count throughout test
- Ramp Up: Gradually increases users from 0 to your set maximum
- Burst Traffic: Simulates varying call volumes (realistic office scenario)
3. Run the Test
Click "Start Test" and watch real-time metrics update. The test will automatically stop after the specified duration, or you can stop it manually.
Understanding VoIP Codecs
Codecs determine how audio is compressed and transmitted. Different codecs use different amounts of bandwidth:
| Codec |
Bandwidth per Call |
Audio Quality |
Best Use Case |
| G.729 |
31.2 kbps |
Good |
Bandwidth-limited networks |
| G.711 |
87.2 kbps |
Excellent |
High-quality audio requirements |
| G.722 |
87.2 kbps |
HD Audio |
Premium call quality |
Note: Bandwidth includes IP/UDP/RTP overhead (approximately 23.2 kbps additional per call)
Understanding the Metrics
Throughput
Maximum: Highest bandwidth usage recorded
Average: Mean bandwidth during test
Minimum: Lowest bandwidth usage
Sessions: Number of active VoIP calls
Call Quality
MOS Score: Mean Opinion Score (1-5, 5 = excellent)
Round Trip: Time for data to travel to endpoint and back
Packet Loss: Percentage of lost data packets
Jitter: Variation in packet arrival times
Device State
Audio: Audio processing capability
Device: Overall device performance
Network: Network adapter status
Bandwidth Speed
Uplink: Upload speed available
Downlink: Download speed available
Best Region: Optimal server location
DNS Lookup
Connected: Successful connections vs. total attempts
Connection Times: How long it takes to establish connections
Pass/Fail Indicators
After each test, metrics are automatically evaluated and marked with indicators:
✓
Pass - Metric meets recommended standards
!
Warning - Metric is borderline, monitor closely
✗
Fail - Metric below recommended standards
Pass/Fail Criteria
- MOS Score: ≥3.5 = Pass, 3.0-3.4 = Warning, <3.0 = Fail
- Packet Loss: <1% = Pass, 1-2% = Warning, >2% = Fail
- Jitter: <20ms = Pass, 20-30ms = Warning, >30ms = Fail
- Round Trip Time: <200ms = Pass, 200-300ms = Warning, >300ms = Fail
- Connection Success: ≥80% = Pass, 60-79% = Warning, <60% = Fail
Interpreting Results
Excellent Performance
All metrics pass standards. Network can reliably support VoIP traffic with high call quality.
Good Performance
Most metrics pass with minor issues. VoIP will work well with occasional quality variations.
Fair Performance
Some metrics show problems. VoIP will work but may experience noticeable quality issues during peak usage.
Poor Performance
Multiple metrics fail standards. VoIP quality will be significantly impacted. Network improvements needed.
⚠️ Quick Troubleshooting Tips
- High Packet Loss: Check network equipment, cables, or contact ISP
- High Jitter: Enable QoS (Quality of Service) on your router
- High Latency: Check for background downloads, streaming, or bandwidth-heavy applications
- Connection Failures: Verify internet connectivity and DNS settings
- Low Bandwidth: Consider upgrading internet plan or reducing concurrent users
Bandwidth Planning
Use these guidelines to estimate bandwidth requirements:
Bandwidth Calculation
Total Required = (Concurrent Calls × Codec Bandwidth) + 20% overhead
Example: 50 concurrent G.729 calls = (50 × 31.2 kbps) × 1.2 = 1,872 kbps (~1.9 Mbps)
Recommended Minimums
- Small Office (1-10 calls): 1-2 Mbps upload/download
- Medium Office (10-50 calls): 5-10 Mbps upload/download
- Large Office (50+ calls): 20+ Mbps upload/download
Best Practices
- Test Regularly: Run tests monthly or after network changes
- Test During Peak Hours: Simulate real-world usage conditions
- Use Realistic User Counts: Test with your actual expected concurrent call volume
- Test Different Codecs: Compare G.729 vs G.711 performance
- Document Results: Keep records for capacity planning and troubleshooting
- Enable QoS: Prioritize VoIP traffic on your network equipment
- Monitor Continuously: Use the burst mode to simulate varying call loads
Technical Notes
This tool simulates VoIP traffic patterns and measures network performance characteristics. It provides estimates based on industry standards and typical VoIP requirements.
Limitations
- Results are estimates based on network simulation
- Actual VoIP performance may vary based on equipment and configuration
- Test uses 8.8.8.8 (Google DNS) as a reliable connectivity endpoint
- Browser-based testing has inherent limitations compared to dedicated tools
For Accurate Production Testing
Consider using dedicated VoIP testing equipment or professional network monitoring tools for production environments with critical VoIP requirements.