What is WebSocket and WebRTC?


A WebSocket is a protocol that allows for two-way communication between a client and a server over a single TCP (Transmission Control Protocol) connection.

It provides a persistent connection between a client and a server, enabling real-time data transfer and communication.

The WebSocket protocol is designed to work over the same ports as HTTP and HTTPS, making it easy to implement and integrate with existing web applications.

It is supported by most modern web browsers and can also be used with server-side technologies such as Node.js, Java, and Python.

Unlike traditional HTTP (Hypertext Transfer Protocol) requests, which are stateless, WebSocket connections are persistent and full-duplex, meaning that both the client and server can send and receive data at any time.

Configure WebSocket in Node Js:

To configure WebSocket in a Node.js project, you can use the “ws” package, which is a simple WebSocket implementation for Node.js.

Here are the basic steps to configure WebSocket in a Node.js project:

npm install ws
const WebSocket = require('ws');
wss.on('connection', function connection(ws) {
  console.log('Client connected');

  ws.on('message', function incoming(message) {
    console.log('Received message: %s', message);

  ws.send('Hello, client!');
wss.listen(8080, function () {
  console.log('WebSocket server started on port 8080');

What is webRTC?

WebRTC (Web Real-Time Communication) is an open-source project that enables peer-to-peer communication between web browsers and mobile applications using standard web technologies.

It allows developers to build real-time communication applications such as video conferencing, file sharing, and online gaming without the need for third-party plugins or software.

WebRTC uses a set of APIs (Application Programming Interfaces) that allow for the establishment of secure, real-time connections between browsers and devices.

It includes several components such as getUserMedia, which allows access to a user’s camera and microphone; RTCPeerConnection, which manages the peer-to-peer connection between browsers; and RTCDataChannel, which enables the exchange of arbitrary data between peers.

Most modern web browsers and mobile devices, including Google Chrome, Mozilla Firefox, Apple Safari, and Microsoft Edge, support WebRTC.

It is also available as a native library for mobile platforms such as Android and iOS.

With its low latency and high-quality audio and video capabilities, WebRTC is ideal for building real-time communication applications that work seamlessly across devices and platforms.

What is PEER in webRTC?

In WebRTC, a peer refers to an endpoint in a real-time communication session that can send and receive audio, video, and data streams.

A WebRTC session typically involves two peers that connect directly with each other, without the need for a centralized server or a dedicated communication channel.

Each peer in a WebRTC session has a unique identity and is responsible for negotiating and establishing a connection with the other peer.

The peers exchange information about their network addresses and capabilities through a process called signaling to establish the connection.

Once the signaling is complete, the peers can use the WebRTC APIs to establish a direct peer-to-peer connection for audio, video, and data transfer.

This allows for real-time communication without the need for intermediaries or server-side processing, making WebRTC a highly efficient and scalable solution for peer-to-peer communication.

In summary, a peer in WebRTC is an endpoint that participates in a real-time communication session and establishes a direct connection with another peer for audio, video, and data transfer.

Read More

Thank You

Related Post

Leave a Reply

Your email address will not be published. Required fields are marked *