VoIP NetworkTester 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.