Measuring quality

VoIP Softswitch with integrated call generator, filter for VoIP wholesale, IVR, looped VoIP traffic and various complex VoIP applications

The software is a Windows-based VoIP softswitch with integrated routing, billing, filtering (fast numbers lookup) and web interface. It can be used for VoIP wholesale carriers. Advanced call processing in the softswitch is based on CallXML scripts and integrated routing/billing engine. The softswitch is able to recognize speech (voice) to text using VOSK speech recognition server (based on Kaldi). The softswitch is a continuously-self-tested system with high stability and performance. We invest efforts into software development and testing, not marketing and web design to keep low prices. Structure of database is clean and simple. SIP and RTP modules (same as in SIP Tester software) are used by more than 800 customers all over the world.

GSM Termination: Free trial on our server Free trial on your server Pricing Tutorial GSM termination guide
Looped generated traffic: Free trial on our server Free trial on your server Pricing Tutorial

Key features

  • Voice-based IVR: recognize speech to text (STT) using VOSK speech recognition server (based on Kaldi).
  • Features for generated VoIP traffic: dialer campaigns, looped generated traffic. See step-by-step tutorial
  • High-performance scalable dialing based on custom CallXML scripts (complex call scenarios)
  • CLI manilupation (replace with numbers from CSV file)
  • Playing a custom ringback tone from WAV file (can be random), can send 180 and 183 to origination side before receiving the ring from termination side. This fake ringback tone reduces PDD that is displayed in customer's CDRs
  • Anti-FAS call processing logic (delay of connection signal)
  • Features for GSM VoIP termination
    • Filtering, test call generator (TCG) robot calls blocking (against SIM block issues). Human/machine detection to reject machine calls, avoid SIMs from blocking in GSM gateways (SIMBOXes), avoid genearated and bad-profile VoIP traffic
      • IVR-based filters (two-stage dialing, "press X to continue the call"). Authentication IVR system to detect calls coming from human by asking to enter a number before send a call to gateway
      • Caller voice based filtering
      • Dynamic blacklists (block A or B number if it is dialed too frequently)
      • Whitelisting, blacklisting with custom complex logic
      • HLR-based filtering
    • CDR Analyser module to build multiple whitelists, blacklists
    • Works with any SIP-compatible VoIP termination route or GSM gateway
    • SIM management features (for GoIP gateways only)
      • Link between CLD (B number) and SIM card (each SIM card can have a dedicated list of dialed numbers)
      • Automatic requests of MSISDN
      • Automatic requests of SIM balance, multi-component balance support
      • Automatic recharging (top-up), various custom methods including conversion to bonus balance component
      • Direct connection from softphone to SIM card, used to manually activate new SIM cards
      • Analysis of blocked SIMs to determine optimal SIM management strategy and settings (for special users and countries only)
      • Intelligent call distribution between SIM cards: daily limits per SIM, per location, min. interval between calls
      • Human behaviour simulation (no details will be published for this feature for security reasons)
      • Supported gateway: Dbltek GoIP
    • Analysis of past traffic and blocked SIMs to determine optimal filtering strategy and settings (for special users and countries only)
    • Detection of ringback tone, call answer and call termination events from RTP audio stream (for SIP-bluetooth-GSM termination with asterisk module chan_mobile)
  • Routing
    • Unlimited originators, routing groups, 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, DID number, tech. prefix
    • Unconditional DID forwarding for VoIP retail, to VoIP provider or SIP phone
    • Black lists, white lists, filtering integration with custom database
    • Custom processing of SIP headers
    • Customizable CallXML scripts for sophisticated call flows (view sample scripts)
  • Billing
    • Prepaid, postpaid
    • Real time balance update: live calls are dropped if balance is out of credit
    • Low balance notification by email
    • Import/export of pricelists from/to CSV files
  • Analysis and reporting
    • Real-time system status dashboard: ASR, ACD, current calls for originators and terminators
    • CDR, ASR/ACD reporting and alerting for originators (customers)
    • CDR reports
      • Basic call information, RTP statistics, recorded file name, custom fields, SIP trace
  • Test call simulator for troubleshooting (to check configuration)
  • 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
    • 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
    • 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
  • Call recording to WAV files and playback in CDR web UI
  • Integration with third-party software, websites: HTTP API, database and API queries in CallXML scripts
  • Topology hiding, NAT traversal (support of originators and terminators behind firewall, using symmetric SIP and RTP)
  • VoIP Protocols: SIP over UDP/TCP/TLS, RTP, RTCP, HTTP
  • Audio codecs: G.711, G.723, G.729. T.38 fax: sending/receiving
  • Open RTP mode: passing RTP packets directly between originator and terminator
  • Operating system: Win7, Win8, Win10, WinServer2008, WinServer2012, WinServer2016, WinServer2019
  • Embedded database and web server for highest performance. Realtime backups of database

Use cases

  • VoIP softswitch for wholesale carrier business
  • SIM management software for GoIP gateways
  • Various VoIP applications: IVR server, conference server, voice mail, virtual attendant, click2call. View sample scripts
  • Call generator (dialer campaigns): generation of VoIP traffic for various purposes, including looped generated traffic. More details here


We have run performance tests with the softswitch on various dedicated servers in order to get servers' call capacity for stable performance, here are results:
Dedicated server, Windows Server 2012 R2 64bit, 4x3.4GHz CPU, 8GB RAM
  • Typical CC traffic, no recording to WAV: ASR = 70%, ACD = 20 seconds, codec = G.729, 70 calls per second: 1300 channels (version 2017-10-09)
  • Typical NCLI traffic, no recording to WAV: ASR = 40%, ACD = 4 minutes, codec = G.729, 14 calls per second: 1200 channels (version 2017-10-09)
VoIP quality indicators (RTP jitter, packet loss, MOS score) have been measured during the tests; the indicators have been OK during the 48-hour load tests.

Pricing for GSM Termination features

Softswitch software license prices

Prices are based on max. number of channels (ports, concurrent calls).
  • Single-server softswitch license for GSM termination (routing, billing, any gateway type via SIP, SIM management for GoIP, standard filters and scripts):
    • 32 channels: 100 USD/month
    • 64 channels: 150 USD/month
    • 128 channels: 250 USD/month
    • unlimited channels: 300 USD/month
  • Single-server softswitch license with custom CallXML scripts (routing, billing, advanced call processing, IVRs, FAS and filtering):
    • 32 channels: 150 USD/month
    • 64 channels: 200 USD/month
    • unlimited channels: 300 USD/month
  • Dialer campaigns and looped generated VoIP traffic: see here
  • Free trial is available here

Technical support

  • Medium-priority technical support: 300 USD monthly
  • High-priority technical support, consulting: 100 USD hourly (prepaid)
  • Softswitch installation and initial configuration: 200 USD
  • Installation of windows on your dedicated server: 200 USD (you can install your self using the tutorial
  • Development of new features: contact us


Startrinity is a highly reliable Softswitch which offers major benefits with a professional support team. Its routing & billing all-in-one functionality is very efficient and works seamlessly. We would definitely recommend it to any start-up & existing business in the wholesale telecom field.
Eyal Astanglov, CTO at Vocalix Ltd

I have been in the business for 21 years I worked with many switches and helped in building some, however this startrinity switch have proven to be one of the most solid switch if not the best when it comes to core operations. It still lacks some functionalities however the owner is working hard to add all needed functions, not only he is brilliant but very dedicated and has an excellent customer service. I would recommend and encourage anyone who is looking for a solid switch. Not to talk about his SIP tester that is been used with one of biggest providers in Canada.

All what I can say, it is simply the best and we should encourge and support such a wonderful work. Thank you for all the hard work and keep up the good work.
Elie Nassar, B Eng., Canada

Since I started to use StarTriniy Softswitch solution I discovered many powerful tools to manage a really professional calls switching business. The routing manipulation and configuration helps a lot for wholesale model and the development team always surprises with new features. Highly recommended
Mario Jara, Paraguay

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

Success stories

Customers #2, #35 and #41 use our softswitch with custom CallXML scripts, complex call processing logic which includes IVR and some other elements. We respect privacy of our clients and don't disclose the details.

Customer #3 has moved away from GoAntiFraud (GAF) and VOS to our softswitch with his GoIP gateways in Africa. The GSM termination is successful and profitable so far with our softswitch and dedicated technical support. We work together on new features in the softswitch and anti SIM blocking solution

Customer #4 has started using the softswitch in 2015 for GSM termination business in Africa. He has implemented custom CallXML scripts for advanced filtering. He has used his own custom SIM management system for GoIP and DINSTAR GSM gateways. Now he is using the softswitch with its billing features for wholesale carrier business too.

Customer #19 from Asia has successfully started VoIP wholesale business with our softswitch. He is trading NCLI routes to Africa, has direct connections with GSM gateway owners (direct routes). The customer helps us with suggestions for development of softswitch, requesting new features and designing new screens

Customer #24 is using the softswitch for retail VoIP origination business in Canada. The VoIP traffic is originated from mobile SIP client or from DID number from Canada to a country in Middle East. Invoices are managed manually via softswitch web interface. We are planning to develop our own mobile dialer app (in 2017) to help this customer with whitelabel bobile app with custom brand. The customer has suggested many reasonable features for the softswitch

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