The software is a free Windows-based class 5 softswitch with billing and web interface. It can be used for VoIP wholesale, origination, termination, also as a platform for building various VoIP applications: PBX systems, IVR servers, conference servers, SBCs, call centers, auto-dialers. Call procesing in the softswitch is based on CallXML scripts. The softswitch is a continuously-self-tested system with high stability and performance. SIP and RTP modules are used by more than 350 customers all over the world.
Key features
Freeware for unlimited channels FAS detection and suppression Generation of FAS: before connection, after disconnection, loopback audio Long PDD detection Detection of dial tone signal in RTP packets, post-dial delay (PDD) measurement. Detection of ringback tone inside RTP packets, reporting of RBT delay Dead air, one-way audio detection RTP jitter, packet loss, low MOS detection Web API for integration with your third-party software Test call generator (TCG) robot calls blocking (against SIM block issues) Detection of ringback tone, call answer and call termination events from RTP audio stream (for SIP-bluetooth-GSM termination with asterisk module chan_mobile) Human/machine detection to reject machine calls or avoid SIMs from blocking in GSM gateways (SIMBOXes) Whitelisting, blacklisting with custom complex logic Google Speech API v2 for automatic speech recognition (ASR) IVRs Continuous VoIP call quality measurement of both caller and called party. Embedded testing of softswitch, IP network, trunks and SIP phones Protocols: SIP over UDP and TCP, RTP, RTCP, HTTP CallXML scripting engine: easy-to-understand, compact scripts, from simple to complex Web interface for configuration Supported audio codecs: G.711, G.723, G.729. Email alerts and reports on SIP trunk call capacity overloads and low audio quality detections
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