Implementing voice calls on a website involves using WebRTC (Web Real-Time Communication) technologies, which allows real-time communication between browsers and devices. Here's a basic guide on how to do it: ## <br>Steps to Implement Simple Voice Calls **1. Configuring the Signaling Server**: - In order for browsers to exchange connection information, a signaling server is required. This can be done with technologies such as WebSockets or libraries such as `Socket.IO`. - The signaling server only serves to facilitate the exchange of information (SDP and ICE candidates) between the peers. **2. Install dependencies**: - Install `Socket.IO` (or other WebSockets technology) on the server. ```bash npm install socket.io ``` **3. Creating the Server**: Set up a `Node.js` server with Express and `Socket.IO`. ```js const express = require('express'); const http = require('http'); const socketIo = require('socket.io'); const app = express(); const server = http.createServer(app); const io = socketIo(server); io.on('connection', (socket) => { console.log('Novo cliente conectado'); socket.on('offer', (data) => { socket.broadcast.emit('offer', data); }); socket.on('answer', (data) => { socket.broadcast.emit('answer', data); }); socket.on('candidate', (data) => { socket.broadcast.emit('candidate', data); }); socket.on('disconnect', () => { console.log('Cliente desconectado'); }); }); server.listen(3000, () => { console.log('Servidor rodando na porta 3000'); }); ``` **4. Creation of the Web Client**: - Use HTML, JavaScript and WebRTC API for the voice call interface. ```html <!-- html code --> <button id="callButton">Chamar</button> <button id="hangupButton">Desligar</button> <audio id="remoteAudio" autoplay></audio> ``` ```js //javascript code const socket = io(); const callButton = document.getElementById('callButton'); const hangupButton = document.getElementById('hangupButton'); const remoteAudio = document.getElementById('remoteAudio'); let localStream; let peerConnection; const configuration = { iceServers: [ { urls: 'stun:stun.l.google.com:19302' } // Servidor STUN público ] }; callButton.onclick = async () => { localStream = await navigator.mediaDevices.getUserMedia({ audio: true }); peerConnection = new RTCPeerConnection(configuration); localStream.getTracks().forEach(track => peerConnection.addTrack(track, localStream)); peerConnection.onicecandidate = event => { if (event.candidate) { socket.emit('candidate', event.candidate); } }; peerConnection.ontrack = event => { remoteAudio.srcObject = event.streams[0]; }; const offer = await peerConnection.createOffer(); await peerConnection.setLocalDescription(offer); socket.emit('offer', offer); }; socket.on('offer', async (offer) => { peerConnection = new RTCPeerConnection(configuration); peerConnection.onicecandidate = event => { if (event.candidate) { socket.emit('candidate', event.candidate); } }; peerConnection.ontrack = event => { remoteAudio.srcObject = event.streams[0]; }; await peerConnection.setRemoteDescription(offer); const answer = await peerConnection.createAnswer(); await peerConnection.setLocalDescription(answer); socket.emit('answer', answer); }); socket.on('answer', (answer) => { peerConnection.setRemoteDescription(answer); }); socket.on('candidate', (candidate) => { peerConnection.addIceCandidate(new RTCIceCandidate(candidate)); }); ``` **Explanation**: - Signaling Server: Facilitates the exchange of connection information (SDP and ICE candidates) between users. - WebRTC API: Manages real-time audio transmission. - STUN Servers: Help discover the public IP address and deal with NATs/firewalls. **Additional requirements**: - TURN server (optional, but recommended to ensure communication on restricted networks). - SSL/TLS certificate, as WebRTC requires secure connections (https). With these steps, you can implement basic voice calls on your website. For production, consider using services like Twilio or other libraries that simplify the process and offer advanced support.