Introduction to WebRTC: What, Why, and How?

What is WebRTC?

WebRTC (Web Real-Time Communication) is an open-source technology that enables real-time communication (RTC) capabilities directly in web browsers and mobile applications. It allows for peer-to-peer audio, video, and data sharing without requiring additional plugins or third-party software.

A Brief History of WebRTC

WebRTC was first introduced by Google in 2011 as an open-source project aimed at simplifying real-time communication. It later gained support from major tech companies and became a W3C standard, allowing developers to build seamless communication experiences directly in web browsers.

Why WebRTC?

The need for WebRTC arose due to the limitations of existing technologies for real-time communication. Traditional methods relied on third-party plugins like Flash and Java applets, introducing security vulnerabilities, performance issues, and compatibility challenges. While WebSockets allowed real-time bidirectional communication, they were not optimized for high-bandwidth applications like video calling. They required a central server for relaying data, leading to increased latency and server costs. WebRTC was developed to solve these issues by enabling direct peer-to-peer connections, reducing latency, improving security, and ensuring seamless real-time communication for applications like video conferencing, live streaming, and online gaming.

WebRTC provides:

• Low Latency: Direct peer-to-peer communication for minimal delay.

• Cross-Platform Support: Works on major browsers like Chrome, Firefox, and Edge.

• Built-in Security: End-to-end encryption for secure data transmission.

• No Plugins Required: Native browser support eliminates the need for Flash or external software.

WebRTC vs WebSockets: When to Use What?

While both WebRTC and WebSockets enable real-time communication, they serve different purposes:

• WebRTC: Designed for peer-to-peer media (audio/video) and data transfer. Ideal for low-latency, high-bandwidth communication, such as video calls and real-time gaming.

• WebSockets: Primarily used for bidirectional communication between a client and a server. Best suited for real-time chat applications, stock updates, and multiplayer turn-based games.

If you need direct peer-to-peer media streaming, WebRTC is the way to go. WebSockets are the better choice if your application requires persistent, reliable communication via a server.

How Does WebRTC Work?

Here, I will provide a high-level overview of how WebRTC functions. In upcoming blogs, I will dive deeper into each of these steps to gain a thorough understanding of the process. WebRTC relies on three main components:

1. Media Stream (getUserMedia): Captures audio and video from the user’s device.

2. Peer-to-Peer Connection (RTCPeerConnection): Establishes direct communication between two clients.

3. Signaling (WebSocket, WebRTC signaling servers): Helps peers discover each other and negotiate connection parameters.

Basic WebRTC Flow:

1. Capture Media: The user’s microphone and camera capture input.

2. Signaling Process: Peers exchange connection details using a signaling server.

3. ICE Candidates Exchange: WebRTC determines the best path for direct communication using STUN/TURN servers.

4. Establish Connection: Peers connect and stream audio, video, or data.

Conclusion

WebRTC is revolutionizing real-time communication by making it accessible, secure, and efficient. In the next blog, I’ll take you through WebRTC’s architecture in detail and break down its key building blocks.