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.

Performance and stability load testing of SIP server

Testing of IVR servers, PBX, call centers and other applications

Bulk call generation

SIP DoS (flood) attack simulation and testing

Audio and fax conformance testing

24x7 monitoring of RTP jitter and VoIP infrastructure availability

IP network assessment

Features

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

SIP/VoIP software developers, system administrators, VoIP integrators, call centers, operators, PBX users