- Make and receive calls via SIP
- Use Turn server for NAT traversal
- Hang up and answer calls
- DTMF support
- Display call status
To run this project, you will need:
- A WebRTC-compatible SIP server
- Credentials for the SIP server
- NodeJs and npm installed on your machine
Clone the repository:
git clone https://github.com/fetristan/webrtc-sip-phone.git
Navigate to the project directory:
cd webrtc-sip-phone
Install dependencies:
npm install
Copy the .env_example file to a new file named .env:
cp .env_example .env
Fill in the .env file with your SIP server and credentials.
To start the application, run:
npm start
The application will be available at http://localhost:3000.
Set up your SIP server and credentials in the .env file:
REACT_APP_SERVER=wss://your-sip-server.com:7443
REACT_APP_URI_STRING=sip:[email protected]
REACT_APP_AUTHORIZATION_PASSWORD=your-password
REACT_APP_AUTHORIZATION_USERNAME=your-username
REACT_APP_TARGET_URI_STRING=sip:[email protected]:5080
REACT_APP_TURN_SERVER=turn:your-turn-server.com:3478
REACT_APP_TURN_SERVER_NAME=your-turn-server.com
REACT_APP_TURN_AUTH_SECRET=your-turn-auth-secret
REACT_APP_TURN_AUTH_EXPIRY=3600
Replace the placeholders with your actual server details.
To make a call to the target, press the '+' button.
To answer an incoming call, press the 'Answer' button when you see an incoming call notification.
To hang up an active call, press the 'Hangup' button.
To use DTMF, press the buttons on the dial pad.
Contributions are welcome! Please feel free to submit a pull request or create an issue for bugs, questions, or new features.
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