callstats-jssip-demo

Yet another example of how WebRTC application using callstats-jssip library, and asterisk SIP as a signalling layer.

Running the example

To run the app, we will need NodeJS and a SIP server. In this example we use Asterisk.

• 1 Clone the repo

  git clone https://github.com/callstats-io/callstats-jssip.git

• 2 Install npm and bower dependencies

  cd  callstats-jssip-demo
  npm install
  bower install    
  

• 3 Provide environments variables either from terminal, or by .env file

        $ ASTERISK_SERVER=csio.testasterisk.io &\ 
           CSIO_APP_ID=81762635 &\ 
           CSIO_APP_SECRET=1213kj1khjkadsghjagsd871 &\
           PORT=9091
  

• 4 Run the demo

  Fill .env file with right credentials

      npm run dev 
  

  or to run without any debug log

      npm start 
  

The demo application is depends on csio JsSIP, and JsSIP library. You can change the library version from codebase before running the application.

Credit

The app is a cloned from https://github.com/agilityfeat/webrtc-sip-example