StarTrinity SIP Tester v3.1
SIP tester is a free VoIP load testing tool which enables you to test VoIP network, SIP software or hardware. It is able to simulate 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.
-Supported operating system: Windows
-Unlimited number of simultaneous calls
-Jitter buffer simulation
-Real time call quality analysis and reports
-RTP jitter, packet loss percentage, answer delay measurement
-Audio recording: mixed and separate RX/TX streams. One can listen recordings and check voice quality
-CDR reports: basic call information RTP statistics
-Least quality calls report: easy to see the worst call in a test
-Supported audio codecs: G.711, G.723, G.729
-T.38 fax support: sending and receiving
-DTMF simulation and analysis: RFC2833 and SIP INFO
-CallXML scripting engine: easy-to-understand, compact scripts, from simple to complex
-Random timer simulation
-Custom WAV file playback
-Bulk generation of SIP calls: manually and on timer
-Receiving calls with and without registration at SIP Tester
-SIP and RTP packet loss simulation
-UAC and UAS registration
-SIP authentication: digest security scheme
- MSSQL integration: saving reports, retrieving list of numbers to call
-Supported specifications: RFC2833, RFC2976, RFC3261, RFC3262, RFC3264, RFC3362, RFC3515, RFC3550, RFC4028
Released:June 25, 2013
Platforms:Windows XP, Windows 2003, Windows Vista, Win7 x32,Win7 x64,WinOther,WinServer
Install:Install and Uninstall
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