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StarTrinity VoIP Status Tutorial

[please note that the tutorial and the software is being updated on the fly as we develop and release new features]
Installation of web interface and database on your Windows server
Deployment of SIP server test node on your Windows server
Running VoIP readiness test
Running continous VoIP test
Detailed technical explanation of VoIP readiness tests

Installation of web interface and database on your Windows server

  • Set up database
    • Set up SQL Server Management Studio (SSMS). There is a free "Express" version
    • Set up MSSQL server. There is a free "Express" version
    • Open MSSQL management studio, connect to server, create database "VoipStatus" (you can choose a different name for the database)
  • Set up web server
    • Go to Server Manager, enable ALL ASP.NET, IIS, web management, WCF features
    • Press Win+R, cmd.exe, run
      %windir%\Microsoft.NET\Framework\v4.0.30319\aspnet_regiis.exe -i
      %windir%\Microsoft.NET\Framework\v4.0.30319\aspnet_regiis.exe -ir
    • Open IIS manager
    • Remove default web site
    • Create new web site. Enter host name (like "voipstatus.yourdomain.com")
    • Download web server binaries from here, unzip, copy into the new website folder. There is a "web.config" file in root of the folder
    • If you have SSL certificate: import the certificate using IIS manager, add HTTPS binding to the website using the imported certificate
    • If you dont have SSL certificate
      • Configure setting "RequireHttps" in web.config file, set to "False":
        <StarTrinity.VoipStatus.Properties.Settings>
         <setting name="RequireHttps" serializeAs="String">
          <value>False</value>
          </setting>
         </StarTrinity.VoipStatus.Properties.Settings>
        </applicationSettings>
    • Configure database connection string in the web.config
      <connectionStrings>
       <add name="VoipStatusDatabase" connectionString="Data Source=localhost\SQLEXPRESS;Initial Catalog=VoipStatus;Integrated Security=SSPI" providerName="System.Data.SqlClient" />
      </connectionStrings>
    • Go to web interface - log in. See database connection error "login failed for user IIS APPPOOL\yourwebsite.com" in the browser
    • In MSSQL management studio create login to connect to the database for the web UI.



      Select "Windows authentication", set "Login name" = "IIS APPPOOL\yourwebsite.com" (from the error in browser)


      Grant rights to the new user


      Add mapping from the new user to the database


      Click OK
  • Log into web interface
    • Default admin credentials: email = "admin", password="123456"
    • Change password (using configuration menu)

Configure company details and settings

  • Edit email (the email is used for login)
  • Edit logo URL. Make sure that the logo URL is accessible in browser
  • Edit company name
  • Edit notifications settings for VoIP readiness tests
  • If you need integration with your ticket system via HTTP API: configure it here

Deployment of SIP server test node on your Windows server

  • Prerequisites
    • If you use web interface and database on our server: account on voipstatus.startrinity.com
    • If you use web interface and database on your server: have it working
    • Windows server for the server-side SIP software
    • Correct time set up on the server
    • Hardware requirements
      • One Amazon EC2 virtual CPU core (2.4GHz) is able to handle 150 concurrent G.711 calls
  • Note: it would be easier if you first install Google Chrome on the server, because IE displays too many security warnings by default
  • Install Wireshark and WinPCAP under admin account, make sure that packet capture works
  • Open ports in windows firewall: UDP 5060..30000 (SIP, RTP), TCP 19019 (SIP Tester web API), TCP 80 (HTTP), TCP 443 (HTTPS)
    • Go to Control Panel - "System and Security" - "Windows Firewall"
    • On left panel click "Advanced settings"
    • Select "Inbound Rules" - click "New rule..."
    • Type of rule = "Port" - Next - "UDP", ports = "5060-30000" - Next - Allow the connection - Name = "SIP and RTP"
    • Add one more rule for TCP port 19019 - Name = "HTTP API for SIP Tester"
  • If you have firewall on hosting provider side: open same ports (for example in Amazon EC2 go to "security groups")
  • In VoIP status web interface go to "Configuration" - "Servers" - "Create new server test node". Enter server's IP address / host name to be used from client side

  • Install SIP tester on your server from MSI installer (zipped StarTrinity.SIPTester.Setup.msi) or ZIP archive with binaries. This should be a dedicated instance of SIP Tester, not used for other tests
  • Run the new installed SIP Tester instance, it will first start in regular mode, not configured for the "VoIP Status" operation
  • On the new server go to browser, log into the web interface - servers - your new server - "Deploy"
  • Click "auto-configure"


  • Check server status in web UI

Running VoIP readiness test

  • Go to Configuration -> Start VoIP readiness test
  • Send download link to the customer. [Customer runs the downloaded exe file]




  • See status of the test, when it is complete see results


    If it rtells you that no calls found for the bandwidth test, make sure that server's clock and time zone are set up correctly
  • If you have issue with the test results, please run VoIP readiness test with our VoIP servers to see if the issue is related to your server side or client side

Running continous VoIP test

  • Go to Configuration - continuous tests - Create new. Enter downtime alert thresholds for the test
  • Go to details of the new test, get "To" user ID, host name, port for further configuration of SIP client device
  • Using Cisco SPA 112 ATA device
    • Connect the device to LAN with DHCP server
    • Get the dynamically assigned IP address of cisco device from your DHCP server. Here is is 192.168.0.13
    • Connect to the device in browser, log in using default credentials "admin"/"admin"
    • In "Quick Setup" enter SIP server details
    • In "Voice" - "SIP" set "RTCP Tx interval" = "1", click "Submit"
    • In "Voice" - "Line 1" - "Proxy and registration" set "Register Expires" = "60", click "Submit"
    • Check LED indicator on cisco ATA device for phone line #1, it must be on, indicating continuous call from the server to the device
  • Check status of the continuous test. It must be "online"
  • See VoIP quality reports in history chart and CDR. Check SIP trace for the calls. Check upstream and downstream jitter and packet loss measurements, round-trip time (RTT)
  • Configure email(s) to receive alerts when the continuous test goes down and up in "Configuration" - "Company details and settings" - "Continuous tests"
  • Turn off the client-side device, make sure that you receive alert by email. Turn it back on, check the "up" notification email. At the same time monitor "Continous tests" webpage (which is automatically refreshed), make sure that test status goes down (red) and up (green)

Detailed technical explanation of VoIP readiness tests

[the chapter is under development]
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