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VoIP Softswitch with integrated billing, filter, SIM management for VoIP wholesale, origination, GSM termination, IVR, IP PBX and various complex VoIP applications

The software is a Windows-based VoIP softswitch with integrated routing, billing, filter, anti SIM blocking solution, SIM management and web interface. It can be used for VoIP wholesale carriers, origination, GSM termination businesses, also as a platform for building various VoIP applications: PBX systems, IVR servers, conference servers, SBCs, call centers, auto-dialers, etc. Call processing in the softswitch is based on CallXML scripts and integrated routing/billing engine. 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 600 customers all over the world.

Free trial on our server Pricing Download MSI Download ZIP Tutorial Change log

Key features

  • Free trial on our server or on your Windows server
  • Flexible prices and dedicated technical support - contact us for details
  • Features for GSM VoIP termination
    • SIM management features
      • 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
      • 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)
    • 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
      • Voice-based filters
      • Dynamic blacklists
      • Whitelisting, blacklisting with custom complex logic
  • 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.
  • 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
  • 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. Preferred: WinServer2012R2 64bit
  • Embedded database for higher performance. Realtime backups of database
  • Putting on hold (RE-INVITE) and transferring (SIP REFER)
  • Termination of international generated traffic
  • Supported specifications: RFC2833, RFC2976, RFC3261, RFC3262, RFC3264, RFC3362, RFC3515, RFC3550, RFC4028

Use cases

  • VoIP softswitch for GSM termination - SIM management, test/spam/trace call filter
  • VoIP softswitch for wholesale carrier business
  • SIM management software for GoIP gateways
  • 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
  • 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

Our services related to the softswitch

We are a technical team with great understanding of SIP, RTP and all other underlying protocols, so we can help you in troubleshooting, if you don't have enough technical skills.
  • Configuring of the softswitch and GSM gateways
  • Identifying reason(s) of low ASR
  • Identifying reason(s) of low ACD
  • Identifying reason(s) of SIM blocking/barring
  • Identifying reason(s) of FAS and over-billing
  • Evaluating patterns of test/trace/junk calls and developing an optimal filter solution
  • Developing complex CallXML scripts for advanced call processing
  • Verification of VoIP route's integrity by sending/receiving traffic from/to our server
Please contact us about prices for the services.


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.


Promotion: the softswitch is free for VoIP wholesalers registered in our VoIP marketplace
Prices are based on maximal number of channels (number of ports / concurrent calls).
  • Softswitch license with support for VoIP wholesale with custom CallXML scripts (advanced call processing and filtering): 2.15 USD/channel monthly (hosted on our server or on your server)
  • Softswitch license without support for VoIP origination: 0.4 USD/channel monthly (hosted on our server or on your server)
  • Softswitch license without support for VoIP wholesale: 0.15 USD/channel monthly (hosted on our server or on your server)
  • Softswitch license without support for GSM termination: 2 USD/channel monthly
  • Softswitch license with support: 5 USD/channel monthly, 20 channels minimum
  • Softswitch license when hosted on our server: 5 USD/channel monthly
  • Softswitch license with priority support: 10 USD/channel monthly, 64 channels minimum
  • Softswitch installation and initial configuration: 100 USD
  • Installation of windows on your dedicated server: 100 USD (you can install your self using the tutorial
  • Fixing of bugs in the softswitch: free
  • Development of new features: contact us (reasonable features could be developed for free)


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


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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