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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