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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.

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: 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

Use cases
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.