- #5.0 NUANCE VOCALIZER FOR NETWORK WINDOWS SERVER 2016 REGISTRATION#
- #5.0 NUANCE VOCALIZER FOR NETWORK WINDOWS SERVER 2016 SOFTWARE#
- #5.0 NUANCE VOCALIZER FOR NETWORK WINDOWS SERVER 2016 PC#
- #5.0 NUANCE VOCALIZER FOR NETWORK WINDOWS SERVER 2016 LICENSE#
- #5.0 NUANCE VOCALIZER FOR NETWORK WINDOWS SERVER 2016 FREE#
False answer supervision (FAS): simulation of fake FAS and detection of FAS for simulated and live calls. SIP authentication: digest security scheme. SRTP (encrypted media): RFC3711, RFC4568. Impairments simulation: SIP and RTP packet loss. #5.0 NUANCE VOCALIZER FOR NETWORK WINDOWS SERVER 2016 REGISTRATION#
Receiving calls with and without registration at SIP Tester. 
Bulk generation of SIP calls: manually and on timer. DTMF generation and detection: RFC2833 and SIP INFO. 
R2S (Real Time Session Supervision Protocol) for ED-137 tests. Support of RTP header extensions for ED-137 air traffic management (ATM) VoIP tests. Supported video codecs: H.264, H.263+, VP8, any other from pcap file. PCAP pass-through mode (RTP from/to PCAP): G.711, G.723, G.729, G.722, GSM, iLBC30, iLBC30, Opus, Silk, Speex, custom Lowest quality calls report: easy to see the worst call in a test. Comprehensive filters allowing searching in CDR database by telephone numbers, qualitative parameters (loss/delay/MOS), codecs CDR reports: basic call information, RTP statistics, recorded file. Audio connection with sound card: speak+listen to simulated SIP calls and check voice quality in real time. One can listen recordings and check voice quality Audio recording: mixed and separate RX/TX streams. Email alarms (alerts) and reports on SIP trunk call capacity overloads and low audio quality detections. Real time reports and charts for every individual call or for all calls. RTP jitter, packet loss percentage, answer delay measurement. Real time analysis of voice quality for all RTP streams, calculation of global min MOS, global max jitter, etc. Jitter buffer simulation with configurable settings. Audio verification: IVR Tests, conference audio path tests (CallXML element verifyaudio). Detection of ringback tone audio signal from received RTP packets post-dial delay (PDD) measurement. Command line interface for SIP unit tests. Support of any user-defined audio/ video codec: RTP playback from. Export of SIP and RTP packets into searete (per call) pcap files. SIPREC: generate SIP calls with 2 RTP streams and play stereo WAV file to test recorders. Custom WAV/MP3/PCAP file RTP audio and video playback. CallXML scripting engine: easy-to-understand, compact scripts, from simple to complex. G.107 E-model and PESQ (P.862.1) MOS and R-factor measurement. 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. High performance: one i7 server can simulate 5400 simultaneous G.729 calls or 2600 G.711 calls (SIP+RTP). Outgoing and incoming SIP calls simulation (active testing, limited in freeware license. #5.0 NUANCE VOCALIZER FOR NETWORK WINDOWS SERVER 2016 FREE#
Free unlimited non-intrusive monitoring of existing VoIP deployment, capturing, recording and analyzing SIP/RTP packets with realtime reports and charts (passive testing, free). #5.0 NUANCE VOCALIZER FOR NETWORK WINDOWS SERVER 2016 LICENSE#
The unlimited license for the SIP Tester is free for non-profit medical organizations (hospitals, research institutes), charity, and nature protection organizations.Īn alternative to SIP Tester for VoIP wholesalers 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.
#5.0 NUANCE VOCALIZER FOR NETWORK WINDOWS SERVER 2016 SOFTWARE#
The software is licensed and protected by law (see license agreement for details).
#5.0 NUANCE VOCALIZER FOR NETWORK WINDOWS SERVER 2016 PC#
The SIP Tester runs on any Windows PC without special hardware and simulates application server, media server, SIP phone or register server.įreeware license of SIP tester allows 50 actively simulated concurrent calls and 150 total attempted+received calls and unlimited passively monitored or recorded calls.įor extended number of calls commercial license is available. 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.Ĭall flow is specified by CallXML script where one can design various situations that can cause failure of tested SIP stack.
StarTrinity SIP Tester™ is a VoIP load testing tool which enables you to test and monitor VoIP network, SIP software or hardware. View more diagrams for different environments
Basic steps for passive testing (monitoring).
0 kommentar(er)
