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VoIP Network Tester

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NetworkTester diagram

Diagram of usage

NetworkTester screenshot

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

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