Diagram of usage
NetworkTester.exe snapshot
VoIP Network Tester
VoIP Network Tester is a free tool which enables you to measure quality of your VoIP network. VoIP call is usually established using a SIP session with a bidirectional RTP stream. SIP and RTP protocols are based on UDP transport protocol. UDP uses a simple transmission model without implicit reliability, ordering and data integrity. Each single UDP packet is transferred independently. The quality of a SIP call depends on delays and loss of IP packets in a network. Long delays lead to large RTP jitter and bad sound quality of a VoIP call. NetworkTester allows you to generate bidirectional UDP streams with a set packet size and bandwidth and measure following:
- Percentage of lost UDP packets
- Maximum and average of jitter time
- Jitter delay distribution view
- Winsockets WSASendTo() procedure execution delays
- VoIP network max bandwidth for a specified jitter
Instructions
- Download the tool on this page
- Launch it on 2 computers
- Optionally specify parameters of UDP streams
- Specify destination IP addresses of peer computer
- Enable senders and receivers
- Run the test and obtain characteristics of your VoIP network. The test can be performed during 24-hours so you will get maximum jitter for this period.
- Configure your VoIP devices according to measured jitter delays and loss percentage
- Choose a better IP provider (with less jitter and packet loss percentage)
- Give a high priority to UDP packets in your IP router