VoIP software

Free Windows-based VoIP Softswitch with integrated billing for VoIP wholesale, origination, GSM termination, IVR, IP PBX and various complex VoIP applications

The software is a free Windows-based class 5 VoIP softswitch with integrated routing, billing and web interface. It can be used for VoIP wholesale carriers, origination, termination businesses, 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. Structure of database is clean and simple. SIP and RTP modules are used by more than 400 customers all over the world.

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Key features

  • Freeware for unlimited channels. There is also commercial license with high-priority technical support - contact us for details
  • Routing
    • Unlimited originators, routes, terminators
    • Algorithms: priority-based / least cost / weight(%)-based (load balancing) / ASR+ACD(quality)-based / prefix-based. Fallback to next route on failure
    • Profit/loss protection
    • Dialed number and CLI manipulation. Normalization of number format
    • Originator authentication based on IP address, user/password, caller ID, PIN code, tech. prefix
    • Originator fraud protection (e.g. against zero-duration call generators)
    • Black lists, white lists, integration with custom database
    • Automatic detection of loops in routes
    • Custom processing of SIP headers
    • Customizable CallXML scripts for sophisticated call flows (view sample scripts)
    • Advanced logic for "missed call" routes (e.g. to Saudi Arabia with zero ASR and ACD): abort call leg B and continue playing fake ring to call leg A - to increase number of channels. The feature is available in commercial license
  • Billing
    • Prepaid, postpaid
    • Real time balance update
    • Generation of invoices
    • 7/3, 7/7, 30/7, other customizable billing cycles
    • Multiple currencies
    • Integration with payment processors
    • Low balance notification by email and/or audio signal during call
    • Import/export of pricelists from/to CSV files
    • Pricelist generator for originators (customers): compare and analyse pricelists of terminators (suppliers), apply margins, generate new pricelist
    • Pricelist update procedure: notification of originators (customers) about changes in price per destination and effective date of new pricelist
  • Analysis and reporting
    • Real-time system status dashboard: ASR, ACD, current calls for originators and terminators
    • Profit/sales reports
    • CDR, ASR/ACD reporting and alerting for originators (customers) and terminators (suppliers)
    • Detection terminators (suppliers) actual capacity based on max. concurrent calls and "503" responses
    • Automatic detection of delays in IP network which result in different billed durations and billing conflicts with originators and terminators
    • CDR reports
      • Basic call information, RTP statistics, recorded file name, custom fields, SIP trace
      • Export to CSV files or database
      • Comprehensive filters allowing searching in CDR database by telephone numbers, qualitative parameters (loss/delay/MOS), codecs
    • Export of SIP and RTP packets into pcap files for individual calls
  • Web interface for administrator, multi-tenant user, originator (calling cards user), terminator (carrier provider) (self-care portal). Customizable logo. Tickets portal for users.
  • WebRTC audio connection for originators and terminators (call via HTML5 browser).
  • Test call simulator for troubleshooting - check if everything is configured correctly
  • FAS detection, suppression, generation
  • Multi-tenancy (usage of single server by multiple independent VoIP business owners, shared hosting)
  • Advanced audio processing and self-testing features
    • Automatic media transcoding for G.711, G.729, G.723 codecs
    • RTP proxy and relay (open RTP route) modes
    • 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
    • Long PDD detection
    • Continuous VoIP call quality measurement of both caller and called party. Embedded testing of softswitch, IP network, trunks and SIP phones
    • Email alerts and reports on SIP trunk call capacity overloads and low audio quality detections
    • RTP jitter, packet loss, low audio quality (MOS) detection
    • Google Speech API v2 for automatic speech recognition (ASR) IVRs
    • Text-to-speech synthesis for IVR prompts (SAPI5)
    • WAV/MP3/PCAP file RTP audio playback
    • DTMF generation and detection: RFC2833 and SIP INFO
    • Recording: mixed and separate RX/TX RTP streams
    • RTP audio signal level measurement
  • Features for GSM VoIP termination
    • 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
  • Web API for integration with third-party software, websites
  • Topology hiding, NAT traversal
  • Protocols: SIP over UDP and TCP, RTP, RTCP, HTTP
  • Audio codecs: G.711, G.723, G.729. T.38 fax: sending/receiving
  • Operating system: Windows XP, Win7, Win8, Win10, WinServer2003, WinServer2008, WinServer2012
  • Embedded database for higher performance. Realtime backups of database
  • Putting on hold (RE-INVITE) and transferring (SIP REFER)
  • Supported specifications: RFC2833, RFC2976, RFC3261, RFC3262, RFC3264, RFC3362, RFC3515, RFC3550, RFC4028

Use cases

  • VoIP softswitch for wholesale carrier business
  • VoIP softswitch for calling card business
    • Pre-recorded IVRs
    • Recharge voucher management
    • PIN or PINless dialing
    • Web API for your website
    • Invoicing
    • Accounts/cards and DIDs management
    • Selection of carier by customer
  • VoIP softswitch for callshop business: unlimited cabins/accounts management, individal cabin/account CDRs. Callback
  • VoIP softswitch for GSM termination
  • Session Border Controller (SBC)
  • PBX system
  • Various VoIP applications: IVR server, conference server, voice mail, virtual attendant, click2call. View sample scripts
  • Call generator (dialer): generation of VoIP calls on schedule

Stable performance: 1500 G.729 channels on 4x3.9GHz CPU

We have tested performance of the softswitch on a windows server. Results are reported here as a table:

Server Test Attempted calls Concurrent calls Avg CPS Codec configuration Softswitch Configuration Max call jitter (avg/90-percentile/max, ms) Max call delta, ms Answer delay, ms Lost packets, %
Intel Core i7-2600@3.4GHz 4 cores, WinServer2012 64bit DTMF, save 2 digits to CDR 260000 240 3,98 G729 ptime=20ms 4 media threads, LWMP=off, PA=on, dtmf=SipInfo, signalDetector=on 7,71/8,2110,33 41,7/45,65/50,48 17/20/111
Intel Core i7-3770@3.9GHz 4 cores, WinServer2012 64bit DTMF, save 2 digits to CDR 1227000 1500 29,94 G729 ptime=20ms 16 media threads, LWMP=off, PA=on, dtmf=SipInfo, signalDetector=off 7,04/7,09/13,73 42,62/39,96/84,42 18,92/23/3526 0,17/0,08/1,04
Intel Core i7-3770@3.9GHz 4 cores, WinServer2012 64bit DTMF, save 2 digits to CDR 1878000 1300 25 G711A ptime=20ms 16 media threads, LWMP=off, PA=on, dtmf=SipInfo, signalDetector=off 8,25/7,75/13,62 35,66/36,75/176,81 13,96/19/559 0,01/0,04/0,32


Softswitch status
Softswitch CDR


The softswitch is free to use by any commercial and non-commercial organizations (freeware). Number of concurrent calls is limited only by hardware resources. There is also commercial license with high-priority technical support - contact us for details


StarTrinity Softswitch would have to be one of the best products I have ever used, the ease of using the CallXML scripting language to deliver the requirements we had to deliver an automated solution to our customers, I would highly recommend this product to anyone exploring.
Steven Sinfield, Australia, Soul Path Psychics

After very long research of VOIP softswitches and trying most of them we found StarTrinity. It has all features we need surprisingly. Thus its developers are very helpful and they welcomed us. Thank you for your all effort, great job!
Cetin Nay, Turkey, SesTeknik


2016-04-26 - lifted our company's priorities for the softswitch, because of getting into 7th place in Google for "SIP Softswitch" request. CTR is very bad, though. It is understandable because target audience expects to see a different thing. We are improving the architecture to make it easy to use for VoIP wholesale and termination businesses: adding new concepts like 'Terminator', 'Originator', 'Balance', 'Tariff'. The softswitch will still be free.
2016-04-30 - we are still taking our time to research competitors' softswitches: VoipSwitch, Kolmisoft MOR and M2, VOS3000, SpeedFlow MediaCore, QoSCalls24, Sippy, PortaOne, MVTS, VoxSwitch
2016-05-21 - defined and implemented entities "Terminator" and "Originator"
2016-05-21 - all further news for Softswitch will be published here
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