VoIP Network Tester is a free UDP flooding and performance testing tool for VoIP networks. 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.