StarTrinity SIP Tester™ is a VoIP load testing tool which enables you to test and monitor VoIP network, SIP software or hardware. It is able to simulate and passively monitor thousands of simultaneous incoming and outgoing SIP calls with RTP media, analyze call quality, and build real-time reports. The call flow is specified by CallXML script where one can design various situations that can cause the failure of tested SIP stack. The SIP Tester runs on any Windows PC without special hardware and simulates application server, media server, SIP phone or register server. Freeware license of SIP tester allows 50 actively simulated concurrent calls and 150 total attempted+received calls and unlimited passively monitored or recorded calls.
For the extended number of calls, a commercial license is available. The software is licensed and protected by law (see the license agreement for details). The unlimited license for the SIP Tester is free for non-profit medical organizations (hospitals, research institutes), charity, and nature protection organizations. Our alternative to SIP Tester is “dialer campaigns” module in StarTrinity Softswitch – you can use it to generate VoIP traffic from our server or your server, using a web interface.

• Free unlimited non-intrusive monitoring of existing VoIP deployment, capturing, recording and analyzing SIP/RTP packets with realtime reports and charts (passive testing, free)
• Outgoing and incoming SIP calls simulation (active testing, limited in freeware license. commercial licenses are available)
• High performance: one i7 server can simulate 5400 simultaneous G.729 calls or 2600 G.711 calls (SIP+RTP)
• Operating system: Windows XP, Windows 7, Windows 8, Windows 10, Windows Server 2003, Windows Server 2008, Windows Server 2012, Windows Server 2016, Windows Server 2019
• G.107 E-model and PESQ (P.862.1) MOS and R-factor measurement
• CallXML scripting engine: easy-to-understand, compact scripts, from simple to complex
• Custom WAV/MP3/PCAP file RTP audio and video playback
• Custom SIP headers and SDP attributes
• Export of SIP and RTP packets into searete (per call) pcap files
• Support of any user-defined audio/video codec: RTP playback from .PCAP file
• Command line interface for SIP unit tests
• Detection of ringback tone audio signal from received RTP packets; post-dial delay (PDD) measurement
• Audio verification: IVR Tests, conference audio path tests (CallXML element verifyaudio)
• Jitter buffer simulation with configurable settings
• Real time analysis of voice quality for all RTP streams, calculation of global min MOS, global max jitter, etc
• RTP jitter, packet loss percentage, answer delay measurement
• Real time reports and charts for every individual call or for all calls
• Email alarms (alerts) and reports on SIP trunk call capacity overloads and low audio quality detections
• Audio recording: mixed and separate RX/TX streams. One can listen recordings and check voice quality
• Audio connection with sound card: speak+listen to simulated SIP calls and check voice quality in real time
• CDR reports: basic call information, RTP statistics, recorded file. Export to CSV files or database. Comprehensive filters allowing searching in CDR database by telephone numbers, qualitative parameters (loss/delay/MOS), codecs
• Lowest quality calls report: easy to see the worst call in a test
• Supported audio codecs: G.711, G.723, G.729.
  PCAP pass-through mode (RTP from/to PCAP): G.711, G.723, G.729, G.722, GSM, iLBC30, iLBC30, Opus, Silk, Speex, custom
• Supported video codecs: H.264, H.263+, VP8, any other from pcap file
• Support of RTP header extensions for ED-137 air traffic management (ATM) VoIP tests. Measurement of round-trip delay
• R2S (Real Time Session Supervision Protocol) for ED-137 tests
• T.38 fax support
• DTMF generation and detection: RFC2833 and SIP INFO
• Bulk generation of SIP calls: manually and on timer
• Receiving calls with and without registration at SIP Tester
• Impairments simulation: SIP and RTP packet loss
• UAC and UAS registration
• SRTP (encrypted media): RFC3711, RFC4568
• SIP over TCP, TLS and UDP transports
• SIP authentication: digest security scheme
• Penetration testing
• False answer supervision (FAS): simulation of fake FAS and detection of FAS for simulated and live calls
• Optional early media for incoming and outgoing calls
• Reading list of destination numbers and SIP accounts from CSV file for both INVITEs and REGISTERs
• Integration with databases via ODBC: saving CDR reports.measurements, retrieving list of numbers for dialing. MSSQL, MySQL, MariaDB, PostgreSQL
• Simulation of putting on hold (RE-INVITE) and transfer (SIP REFER)
• Loopback audio connection(between RX and TX RTP audio streams)
• RTP audio signal level measurement in dB
• Free VoIP recording: capturing G.711 and G.729 RTP streams into WAV files via mirroring port, saving CDRs to your database via ODBC driver
• Supported specifications: RFC2833,

Routing:

• Least Cost Routing
• Load balancing/sharing for termination endpoints
• Traffic failover with multiple “waterfall” levels
• Multiple routing plans

Routing:

• Least Cost Routing
• Load balancing/sharing for termination endpoints
• Traffic failover with multiple “waterfall” levels
• Multiple routing plans