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WHOLESALE CARRIERS, ORIGINATION, GSM TERMINATION

StarTrinitySoftswitch

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

StarTrinitySoftswitch

VoIP Softswitch with integrated billing, filter, SIM management for VoIP wholesale, origination, GSM termination, IVR, IP PBX and various complex VoIP applications
https://vpsvos.com/wp-content/uploads/2020/05/header-bg01.png
SIP TESTER, CONTINUES SPEED TEST, SIP SOFTSWITCH

Develops World Class
Software for VoIP Systems

StarTrinity software is based on Microsoft Windows, C#, .NET, WPF, HTML, javascript and C/C++. Development methods

  • Overview
  • Key features
  • Use cases
  • User Portal

StarTrinity SIP Tester™ is a VoIP load testing tool which enables you to test and monitor VoIP network, SIP software or hardware. It is able to simulate and passively monitor thousands of simultaneous incoming and outgoing SIP calls with RTP media, analyze call quality, and build real-time reports. The call flow is specified by CallXML script where one can design various situations that can cause the failure of tested SIP stack. The SIP Tester runs on any Windows PC without special hardware and simulates application server, media server, SIP phone or register server. Freeware license of SIP tester allows 50 actively simulated concurrent calls and 150 total attempted+received calls and unlimited passively monitored or recorded calls.
For the extended number of calls, a commercial license is available. The software is licensed and protected by law (see the license agreement for details). The unlimited license for the SIP Tester is free for non-profit medical organizations (hospitals, research institutes), charity, and nature protection organizations. Our alternative to SIP Tester is “dialer campaigns” module in StarTrinity Softswitch – you can use it to generate VoIP traffic from our server or your server, using a web interface.

  • Free unlimited non-intrusive monitoring of existing VoIP deployment, capturing, recording and analyzing SIP/RTP packets with realtime reports and charts (passive testing, free)
  • Outgoing and incoming SIP calls simulation (active testing, limited in freeware license. commercial licenses are available)
  • High performance: one i7 server can simulate 5400 simultaneous G.729 calls or 2600 G.711 calls (SIP+RTP)
  • Operating system: Windows XP, Windows 7, Windows 8, Windows 10, Windows Server 2003, Windows Server 2008, Windows Server 2012, Windows Server 2016, Windows Server 2019
  • G.107 E-model and PESQ (P.862.1) MOS and R-factor measurement
  • CallXML scripting engine: easy-to-understand, compact scripts, from simple to complex
  • Custom WAV/MP3/PCAP file RTP audio and video playback
  • Custom SIP headers and SDP attributes
  • Export of SIP and RTP packets into searete (per call) pcap files
  • Support of any user-defined audio/video codec: RTP playback from .PCAP file
  • Command line interface for SIP unit tests
  • Detection of ringback tone audio signal from received RTP packets; post-dial delay (PDD) measurement
  • Audio verification: IVR Tests, conference audio path tests (CallXML element verifyaudio)
  • Jitter buffer simulation with configurable settings
  • Real time analysis of voice quality for all RTP streams, calculation of global min MOS, global max jitter, etc
  • RTP jitter, packet loss percentage, answer delay measurement
  • Real time reports and charts for every individual call or for all calls
  • Email alarms (alerts) and reports on SIP trunk call capacity overloads and low audio quality detections
  • Audio recording: mixed and separate RX/TX streams. One can listen recordings and check voice quality
  • Audio connection with sound card: speak+listen to simulated SIP calls and check voice quality in real time
  • CDR reports: basic call information, RTP statistics, recorded file. Export to CSV files or database. Comprehensive filters allowing searching in CDR database by telephone numbers, qualitative parameters (loss/delay/MOS), codecs
  • Lowest quality calls report: easy to see the worst call in a test
  • Supported audio codecs: G.711, G.723, G.729.
    PCAP pass-through mode (RTP from/to PCAP): G.711, G.723, G.729, G.722, GSM, iLBC30, iLBC30, Opus, Silk, Speex, custom
  • Supported video codecs: H.264, H.263+, VP8, any other from pcap file
  • Support of RTP header extensions for ED-137 air traffic management (ATM) VoIP tests. Measurement of round-trip delay
  • R2S (Real Time Session Supervision Protocol) for ED-137 tests
  • T.38 fax support
  • DTMF generation and detection: RFC2833 and SIP INFO
  • Bulk generation of SIP calls: manually and on timer
  • Receiving calls with and without registration at SIP Tester
  • Impairments simulation: SIP and RTP packet loss
  • UAC and UAS registration
  • SRTP (encrypted media): RFC3711, RFC4568
  • SIP over TCP, TLS and UDP transports
  • SIP authentication: digest security scheme
  • Penetration testing
  • False answer supervision (FAS): simulation of fake FAS and detection of FAS for simulated and live calls
  • Optional early media for incoming and outgoing calls
  • Reading list of destination numbers and SIP accounts from CSV file for both INVITEs and REGISTERs
  • Integration with databases via ODBC: saving CDR reports.measurements, retrieving list of numbers for dialing. MSSQL, MySQL, MariaDB, PostgreSQL
  • Simulation of putting on hold (RE-INVITE) and transfer (SIP REFER)
  • Loopback audio connection(between RX and TX RTP audio streams)
  • RTP audio signal level measurement in dB
  • Free VoIP recording: capturing G.711 and G.729 RTP streams into WAV files via mirroring port, saving CDRs to your database via ODBC driver
  • Supported specifications: RFC2833,

Routing:

  • Least Cost Routing
  • Load balancing/sharing for termination endpoints
  • Traffic failover with multiple “waterfall” levels
  • Multiple routing plans

Routing:

  • Least Cost Routing
  • Load balancing/sharing for termination endpoints
  • Traffic failover with multiple “waterfall” levels
  • Multiple routing plans
StarTrinity

Mobile VoIP

Mobile VoIP with Rich Communication Suite for smartphones and tablets.

Callback & Callshops

Cutting edge retail services from Calling cards through Callback to Callshops.

Reporting

Charts reports for concurrent calls, call duration and ASR with sales reports for payments, cost, and revenue

Multilevel Reseller

Multilayered ownership structure to support your reseller/partner channels and networks of sales agents.

Our Featured
Dedicated Servers

$49
2 Core Dedicated Server
Intel® Xeon E3-1270 v3 2.40 GHz
  • 4 GB RAM
  • 250 GB SCSI Hard Drives
  • Internet Port 100 Mbps
  • Bandwidth 5000 GB
  • 24x7 Monitoring System
  • Network Uptime SLA - 99.999%
$99
4 Core Dedicated Server
Intel® Xeon E5-2620 v3 2.50 GHz
  • 8 GB RAM
  • 350 GB SCSI Hard Drives
  • Internet Port 100 Mbps
  • Bandwidth 10 TB
  • 24x7 Monitoring System
  • Network Uptime SLA - 99.999%
$129
8 Core Dedicated Server
Intel® Xeon E5-2630 v3 2.60 GHz
  • 16 GB RAM
  • 500 GB SCSI Hard Drives
  • Internet Port 100 Mbps
  • Bandwidth 20 TB
  • 24x7 Monitoring System
  • Network Uptime SLA - 99.999%
$149
8 Core Dedicated Server
Intel® Xeon E5-2630 v3 2.60 GHz
  • 16 GB RAM
  • 500 GB SCSI Hard Drives
  • Internet Port 1 Gbps
  • Bandwidth 50 TB
  • 24x7 Monitoring System
  • Network Uptime SLA - 99.999%
$179
12 Core Dedicated Server
Intel® Xeon E5-2620 v3 2.50 GHz
  • 32 GB RAM
  • 500 GB SCSI Hard Drives
  • Internet Port 1Gbps
  • Bandwidth 50 TB
  • 24x7 Monitoring System
  • Network Uptime SLA - 99.999%
$380
16 Core Dedicated Server
Intel® Xeon E5-2630 v3
  • 64 GB RAM
  • 500 GB SCSI Hard Drives
  • Internet Port 100 Mbps
  • Bandwidth 20 TB
  • 24x7 Monitoring System
  • Network Uptime SLA - 99.999%
voip support

GENERAL INFORMATION ABOUT VOIPSWITHWhat does
Voipswitch do?

Voipswitch Class 4/5 Softswitch is a multiservice telephony server that leverages SIP and other related communication protocols.
Can I rent the StarTrinity Softswitch?

Yes, we offer both to rent and to onetime installation to your server.

Do you offer trainings?

Yes, we give training and others tutorial documentation with videos. Usually we provide an online session but also when requested we provide onsite training, it can be in our offices or at customer.

Can I host the softswitch on your server?

Yes, we work with the datacenter in USA and Europe an can offer you a full of package including server, backup, IP addresses etc. If you are asian, we have high redundant singapore server also we care about terminated and call genaration server location which you get better call quality to fine tune. We can suggest you which colocation is better for your termination.

Can SIP Tester run on Linux OS?

Sorry, no. We use C#, .NET, WPF, and some other Microsoft technologies which make the development more effective. We may support Linux in the future.

Which programming language is used in SIP Tester?

SIP Tester is developed with C# (GUI and high-level logic), C/C++ (SIP and RTP processing), XAML, HTML, and Javascript. Some procedures like memcpy are written in ASM.

How to run automated tests with SIP Tester? Is there any API?
There is a command-line interface (CLI), you can use it for automated tests. There are also “readfile”, “writefile”, “writecdr” CallXML elements for advanced API. The results of testing could be exported to CSV files or databases. In addition to that, there is a web API that is used by our AJAX-based web interface.
TOTALLY SERVED VOIPSWITCH TO CLIENT

Why it's worth
to choose VPSVOS?

Our successfully delivered services to give end-users where they get full of satisfied with us.
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Installation provided third party server

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Hosted solution provided

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Countries we served

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CONTACT US WITH EASEGet in touch

Visit our company or simply send us an email anytime you want. If you have any questions, please feel free to contact us.
Address
19854 Michigan Avenue Brekford,
CA 599780 USA.
Call us
USA : +134 6340 0004
Toll :+1 888 571 3448
Email us
sales@vpsvos.com
info@vpsvos.com
support@vpsvos.com
MORE THAN JUST BUSINESS

We Are Trusted
By Industry Leaders

Our trade references from where we got the badge for trust VoIP business solution provider in the next-generation marketplace.
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VPSVOS COMMUNICATIONHeadquarters
VPSVOS Communications is a registered company in the United States of America. Registered Office: 19854 Michigan Avenue Brekford, CA 599780 USA.
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VPSVOS COMMUNICATIONHeadquarters
VPSVOS Communications is a registered company in the United States of America. Registered Office: 19854 Michigan Avenue Brekford, CA 599780 USA.
OUR LOCATIONSWhere to find us?
https://vpsvos.com/wp-content/uploads/2019/04/img-footer-map.png
FOLLOW USVPSVOS Social Connect
Stay up to date with the latest news about VPSVOS Communication, events, special offers and more by connecting with our social media.

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