SIP Tester stressing tool
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Documentation
Sample CallXML scripts
SIP tester is a free load testing software which enables you to run stressing and performance tests for your SIP hardware or software.
It generates and receives many SIP calls simultaneously.
Call flow is specified by CallXML-like script where you can design many various situations which can cause failure of SIP hardware or software which is being tested.
For example you can create and abort call immediately, make 100 calls in a second, send multiple DTMF (SIP INFO), REFER, RE-INVITE commands within a call.
Also you can record and play audio, send and receive faxes and check recordings after test in order to check the quality of jitter buffer and IP network.
The tool is compatible with RFC3261 specification. It has been tested with Audiocodes Mediant, AudioCodes MPxxx, LinkSys,
Fastwire OpenCA, VocalTec Essentra CX, Planet VIP-28x, Addpac AP200D, Telesis Stilllink and many others.
Features
- Performance and stability testing of SIP-based VoIP network components
- Bulk call generation (500 concurrent calls on an average PC)
- Windows XP, Windows 7, Windows Server 2003, Windows Server 2008
- CallXML-like scripts for tests
- Random tests
- Receiving calls with and without registration at SIP Tester
- Registration at SIP server
- Send/receive DTMF
- Play/record audio messages
- Send/receive T.38 faxes
- SIP DoS simulation
- Monitoring of SIP infrastructure
Screenshot
If you have any questions please contact us