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, etc. 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