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SIP tester is a VoIP load/stress testing and monitoring tool which enables you to test VoIP networks, SIP software or hardware. It is able to simulate and monitor thousands of simultaneous incoming and outgoing SIP calls with RTP media, analyze call quality and build real time reports. Call flow is specified by CallXML script where one can design various situations which can cause failure of SIP stack which is being tested. The SIP tester runs on any Windows PC without special hardware and simulates application server, media server, SIP phone or registrar. Features Freeware unlimited non-intrusive monitoring of existing VoIP deployment, capturing and analyzing SIP/RTP packets with real-time reports and charts (passive testing, free) High performance: one i7 server can simulate 5400 simultaneous G.729 calls or 2600 G.711 calls Scalable distributed architecture with SIP and RTP modules modularly installed in one or many PCs Audio verification: identification of IVR messages, conference testing Real time analysis of voice quality for all RTP streams, calculation of global min MOS, global max jitter, etc Lowest quality calls report: easy to see the worst call in a test Support of RTP header extensions for ED-137 air traffic management (ATM) VoIP tests Measurement of round-trip delay VoIP recording ...etc Use cases Performance and stability load testing of SIP servers and IP networks. Testing of memory leaks with millions of calls. Testing of IVR servers, PBX, call centers and other applications when upgrading to new version or moving to new SIP platform Monitoring VoIP quality of existing IP networks and servers, generating alerts and reports about system's performance: high jitter or packet loss, outage, reach of trunk's capaciuty limit Testing and monitoring of SIP-GSM, SIP-PSTN gateways, SIP trunks 24x7 monitoring of RTP jitter and VoIP infrastructure availability ...etc