WebRTC (Web Real-Time Communication) is a free and open-source project providing web browsers and mobile applications with real-time communication (RTC) via application programming interfaces (APIs). It allows audio and video communication to work inside web pages by allowing direct peer-to-peer communication, eliminating the need to install plugins or download native apps.Supported by Apple, Google, Microsoft, Mozilla, and Opera, WebRTC specifications have been published by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF).According to the webrtc.org website, the purpose of the project is to "enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols".

Wikipedia

Overview

WebRTC, or Web Real-Time Communications, is an open-source project that provides real-time voice, text, and video communication capabilities between web browsers and devices. It uses application programming interfaces (APIs) to create peer-to-peer (P2P) communications between internet web browsers, enabling real-time audio and video communication simply by opening a webpage^[1].

WebRTC is designed to solve the problem of real-time P2P communications over the internet. It is used for video chats and meetings on platforms like Zoom, Microsoft Teams, Slack, or Google Meet. Industries such as healthcare, surveillance and monitoring, and the Internet of Things (IoT) also use WebRTC.
The data transfer occurs in real time without the need for custom interfaces, extra plugins, or special software for browser integration^[1:1].

WebRTC uses JavaScript, APIs, and Hypertext Markup Language (HTML) to embed communication technologies within web browsers. It works with most major web browsers and its APIs perform several key functions, including the acquisition of media, creation of peer connections, and data communication^[1:2][2].

Connection Flow

• A wants to connect to B
• A finds out all possible ways the public can connect to it
• B finds out all possible ways the public can connect to it
• A and B signal this session information via other means like WhatsApp, QR, Tweet, WebSockets, HTTP Fetch..
• A connects to B via the most optimal path
• A & B also exchanges their supported media and security

Pros

1. Open Source: WebRTC is an open-source project, which means it is freely available for developers to use and modify^[3].
2. Security: WebRTC provides a high level of security against external attacks^[3:1].
3. Quality: It offers better sound quality due to the use of G.711 and Opus codecs for transferring audio^[3:2].
4. Compatibility: WebRTC is supported by most leading web browsers^[3:3].
5. Real-time communication: It enables real-time audio, video, and data communication without the need for plugins or third-party software^[1:3].

Cons

1. Still under development: WebRTC is still under development as a communication standard, which can lead to some challenges^[4].
2. Dependence on Internet Connection: The quality of the broadcast can be limited by the viewer's internet connection speed^[5].
3. Not suitable for large events: WebRTC may not be suitable for streaming large events with high network load^[5:1].
4. Interoperability: There is no standard signaling protocol, which means there is no guarantee that one WebRTC application can communicate with another^[6].

In conclusion, WebRTC is a powerful tool for enabling real-time communication on the web. It has several advantages, including being open-source, secure, and offering high-quality audio and video communication. However, it also has some limitations, such as being still under development and having potential issues with internet connection speed and interoperability.

Also Read

• 59.15 gRPC
• 59.34 WebSockets
• 59.25 OSI Model