What is WebRTC: A Comprehensive Guide to Real-Time Communication on the Web

When engaging in audio and video calls, screen sharing, or real-time data transfer through your web browser, you may not often ponder the underlying technology. However, one of the core driving forces behind these functionalities is WebRTC. Announced by Google in 2011, this open-source project now has developed a highly comprehensive and continuously expanding ecosystem.

With just a bit of coding knowledge, anyone can now create sophisticated applications for communication, collaboration, and data-rich experiences in hours or days. WebRTC is changing the tech landscape, empowering aspiring developers and creators.

Unveiling the Core of WebRTC

WebRTC is an open-source project that empowers real-time communication directly within web browsers. It eliminates the need for additional plugins or downloads, providing a seamless experience for users. The project offers a set of APIs and protocols that create direct peer-to-peer(P2P) communication to allow secure audio and video communication and data sharing between browsers.

We can break it down into 3 points:

• It's a technology that offers low-latency, high-quality real-time audio and video communication.
• It's a comprehensive client-side multimedia framework with audio and video processing capabilities, designed for cross-platform compatibility.
• It's a standardized set of APIs (part of the W3C recommendation), allowing web developers to create diverse real-time audio and video applications.

WebRTC enjoys support from all major web browsers (Google Chrome, Mozilla Firefox, Safari), making it a universal solution for real-time communication.

How does WebRTC work?

WebRTC offers a user-friendly JavaScript API layer that can be utilized within web browsers. This significantly simplifies the development and integration of real-time communication features across various applications. While WebRTC's core implementation primarily relies on C/C++, most developers using WebRTC won't have to delve deeply into these lower-level details to create their applications.

The WebRTC architecture is characterized by a layered and modular structure.

The Web APP layer utilize the standard Web API interfaces provided by WebRTC for real-time audio and video communication. It is applications developed by developers based on the WebRTC technology specification. Strictly speaking, this doesn't fall under the architecture of WebRTC itself.

The WebRTC Native C++ API layer is provided by WebRTC for application layer developers. At this level, developers don't need to delve into intricate underlying technical details. They only need to have a general understanding of WebRTC's workflow and principles to utilize its APIs for implementing peer-to-peer communication features.

The top layer of the WebRTC core comprises the Transport / Session layer, responsible for managing logic related to audio and video data transmission and processing. It involves various protocols, such as SDP (Session Description Protocol) for signaling servers, mainly used for signaling exchange and managing the connection state of RTCPeerConnection.

The essential components of WebRTC include the video engine, audio engine, and transport module. They have the responsibility of collecting and recording data from devices in video, audio, and text formats. These three modules work together to enable real-time communication and data transfer within the WebRTC framework.