VoIP and SMS Softswitch
The StarTrinity Softswitch is a comprehensive solution designed to optimize your operations and maximize efficiency with VoIP and SMS.
The softswitch includes VoIP (SIP) and SMS (SMPP) modules with CallXML scripting engine, web interface and API.
It is a flexible continuously-self-tested system with high stability and performance.
We make efforts on the software development and testing instead of marketing and web design.
Structure of database is clean and simple.
SIP and RTP modules (same as in SIP Tester software)
are used by more than 1000 customers all over the world.
Our software is currently trusted by numerous VoIP and SMS partners who report significant improvements in their operations, cost savings, and revenue generation.
The softswitch offers a wide range of advanced features tailored to meet the needs of telecom providers:
How to use
Rich set of features
- Looped generated VoIP and SMS traffic: send and receive generated traffic at same IP
- Business purposes: generate revenue, move money, manipulate balance with partner, optimize taxes, invoices, make deals with partners.
Exploit price (rate) differences to gain profit by running generated VoIP traffic via interconnected VoIP partners
- A/B whitelists, blacklists, dynamic blacklists: avoid repetitive numbers/prefixes: make generated traffic look like real traffic
- Volume limit per destination
- Configurable daily call volume pattern for generated traffic: make look like real traffic
- Read A/B numbers from CSV or generate randomly from mask
- API to upload numbers from your database, start/stop campaigns, export CDR
- Control ACD, ASR, SIP codes, ringback tones:
- Send 180/183 to orig. side before receiving the ring from term. side: reduce PDD that is displayed in customer's CDRs
- Control SMS DLR behavior for SMS receiver
- Various techniques to hide generated VoIP and SMS traffic (contact us for details)
- Play recorded WAV files like real voice calls
- Highly customizable CallXML scripting engine for complex applications
- Cancel VoIP call or stop SMS generator when traffic goes to another route
- Play recordings from numerous WAV files (for IVR)
- Termination to IVR (script-based) or to SIP phones (like call center)
- Traffic filtering for VoIP wholesale and GSM termination: improve ACD/ASR and to reduce SIM blocking
- Works with any SIP-compatible VoIP termination route or GSM gateway
- A/B prefix-based whitelisting
- A/B prefix-based blacklisting
- Dynamic blacklisting: prevent repetitive A and B numbers. Limit daily connected time and attempts per prefix
- Delay with fake ringback tone to filter out short-duration cancelled calls
- HLR-based filtering
- 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
- Anti-FAS call processing logic (delay of connection signal)
- CDR Analyser module to build multiple whitelists, blacklists
- Detection of ringback tone, call answer and call termination events from RTP audio stream (for SIP-bluetooth-GSM termination with asterisk module chan_mobile)
- 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
- 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
- Originator authentication based on IP address, user/password, caller ID, PIN code, DID number, tech. prefix
- Basic 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
- Features for traffic manipulation (for test purposes)
- CLI manipulation: replace with numbers from CSV file
- SIP codes mapping: change SIP status code when transferring from terminator to originator
- Simulate FAS: play WAV files as IVR + leave hanging call after X connected seconds with WAV file. See sample scripts here: FAS detection, suppression, generation
- 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
- Simulate fake SMPP DLRs
- Custom SIP headers processing
- Customizable CallXML scripts for sophisticated call flows (view sample scripts)
- Web interface and API: CDR, originators, terminators, routing, basic billing, campaigns, live calls, lists, SMPP configuration and SMS CDR
- Test call simulator for troubleshooting (to check configuration)
- 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)
- 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)
- Supported Protocols: SIP over UDP/TCP/TLS, RTP, RTCP, HTTP, SMTP
- 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
- Call generator (dialer campaigns): generation of VoIP traffic for various purposes, including looped generated traffic. Step-by-step tutorial is here
- SMS generator (SMS campaigns): generation of SMS traffic, including looped generated traffic
- 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
Documentation
Performance
Reviews
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