Overview

A full featured Webphone using JsSIP.

• 4 Lines for Call
• Easy to use and powerful API of JsSIP library
• SIP over WebSocket
• Lightweight!
• Feature like Mute, Hold, Transfer and DND (Do not Call)
• Attractive UI

Technology Used

• JsSIP, the JavaScript SIP library
• React + Vite
• React Icons
• Tailwindcss

Docker Support

This project comes with Docker support, making it easy to deploy the Webphone in a containerized environment.

Prerequisites

• Docker installed on your machine

Setup and Running with Docker

1. Pull the Docker image from Docker Hub:

docker pull aghasuhail96/webphone

2. Run the container:

docker run -p 80:80 aghasuhail96/webphone

3. Access the Webphone app at http://localhost:80.

Usage

const phone = new PhoneLib(SIP_USER, SIP_PASS, URL_WEBRTC, PORT_NO_WEBRTC)

//Connect to the SIP Server
phone.connect()

Events

//On Incoming Call
phone.on('incoming', (data)=>{
  console.log(data)
})

//On Outgoing Call
phone.on('outgoing', (data)=>{
  console.log(data)
})

//On Line Change
phone.on('changeLine', (data)=>{
  console.log(data)
})

//On Call Transfered
phone.on('transfer', (data)=>{
  console.log(data)
})

//On Calls Rejected while DND is enable
phone.on('dndRejected', (data)=>{
  console.log(data)
})

//On Line Hold
phone.on('hold', (data)=>{
  console.log(data)
})

//On Line UnHold
phone.on('unhold', (data)=>{
  console.log(data)
})

//On Line Muted
phone.on('mute', (data)=>{
  console.log(data)
})

//On Line UnMuted
phone.on('unmute', (data)=>{
  console.log(data)
})

Contributing

Pull requests are welcome. For major changes, please open an issue first to discuss what you would like to change. Please make sure to update tests as appropriate.

Screenshot