Icon of program: StarTrinity Softswitch

StarTrinity Softswitch for Windows

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StarTrinity Softswitch Analysis

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StarTrinity Softswitch is a free VoIP software designed for Windows that facilitates advanced communication solutions. This application serves as a softswitch, enabling users to manage and route voice calls over the internet seamlessly. It supports various protocols and provides features such as call routing, billing, and real-time monitoring, making it suitable for both personal and business use.

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The software is equipped with an intuitive interface that simplifies the management of VoIP services. Users can configure multiple gateways and set up comprehensive call handling rules. With its focus on flexibility and scalability, StarTrinity Softswitch is ideal for telecommunications providers and organizations looking to enhance their communication infrastructure.

Icon of program: StarTrinity Softswitch

StarTrinity Softswitch for Windows

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Used StarTrinity Softswitch for Windows?

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Full Specifications

GENERAL
Release
Latest update
Version
3.1
OPERATING SYSTEMS
Platform
Windows
Operating System
  • Windows Vista
  • Windows 7
  • Windows 8
  • Windows XP
  • Windows Server
  • Windows 10
  • Windows 2003
Additional Requirements
None
POPULARITY
Total Downloads
15
Downloads Last Week
0
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Last Updated

Developer’s Description

By StarTrinity
Use a soft switch with billing and Web interface in your VoIP infrastructure.
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 channelsFAS detection and suppressionGeneration of FAS: before connection, after disconnection, loopback audioLong PDD detectionDetection of dial tone signal in RTP packets, post-dial delay (PDD) measurement. Detection of ringback tone inside RTP packets, reporting of RBT delayDead air, one-way audio detectionRTP jitter, packet loss, low MOS detectionWeb API for integration with your third-party softwareTest 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 logicGoogle Speech API v2 for automatic speech recognition (ASR) IVRsContinuous VoIP call quality measurement of both caller and called party. Embedded testing of softswitch, IP network, trunks and SIP phonesProtocols: SIP over UDP and TCP, RTP, RTCP, HTTPCallXML scripting engine: easy-to-understand, compact scripts, from simple to complexWeb interface for configurationSupported 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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AI Assisted Content Disclosure

Content created and reviewed by Softonic with information obtained from StarTrinity, using AI.

CNET's editorial team was not involved in the creation of this content. Opinions, analysis and reviews were not provided by CNET.