This repo contains a simple example of how to build a WebRTC application usign SIP as signaling layer.
The example by no means represents a production-ready application nor presents secure practices.
The main objective is to show what would be the workflow in a WebRTC app tha uses SIP for signaling.
To run the app, you will need NodeJS and a SIP server. In this example we use Asterisk.
After cloning the repository, open js/main.js and set the domain variable to your server address.
Then install the npm dependencies an run the application with npm start. The app will be available at https://localhost:8080
git clone https://github.com/agilityfeat/webrtc-sip-example.git
cd webrtc-sip-example
npm install
npm start
Open the app in two separate tabs, log in as users bob and lucy, and click Call. Note that an insecure password has been set and you definitely don't want to do this in an production environment.
For the example to work you need to set up an Asterisk Server. We used version 15.
Configuration files used in this example can be found in the asterisk-conf folder.
- We use http-server to serve the files.
- We use jssip for sip support
- We use sdp-interop-sl for making some required adjustments to sdp in chrome