What is WebRTC and How Does It Work?

The recent pandemic forced businesses to adopt the remote working model, and while email was the preferred communication channel, the communication wasn’t happening in real-time. This increased the demand for alternative business solutions for ensuring team communication in real-time without lag. So, businesses started looking for VoIP-based business communication solutions that allow conferencing as well as support rich media.

What is WebRTC?

WebRTC (Web Real-Time Communication) is a powerful VoIP technology enabling seamless peer-to-peer communication between web browsers and applications. Application programming interfaces (APIs) written in JavaScript make it easy for software developers to integrate different features using WebRTC. These APIs can be included in the developers’ applications in order to enable peer-to-peer (P2P) communications. There are no compatibility issues between the web and mobile applications, and there is audio, video, and text-based content support.

WebRTC enables real-time data flow within web browsers, so developers don’t need to rely on proprietary interfaces, additional plugins, or specialized software for browser integration. Real-time audio and video communication is as simple as opening a webpage and initiating a call.

Understanding WebRTC

WebRTC is an open-source project initiated by Google in 2011, aiming to provide a standardized framework for real-time communication within web browsers. It is built on a set of JavaScript APIs and protocols that allow direct peer-to-peer communication without requiring any additional plugins or software installations. WebRTC encompasses audio, video, and data channels, making it versatile for various communication needs.

Key Applications using WebRTC technology are:

• Google Meet
• Facebook Messenger
• Discord
• Amazon Chime
• Houseparty
• GoToMeeting

Key Components of WebRTC:

To grasp the inner workings of WebRTC, it is essential to understand its core components:

1. MediaStream: MediaStream represents the audio and video streams captured from the user’s microphone and camera, facilitating real-time communication.
   
   
2. RTCPeerConnection: This API enables the establishment of direct peer-to-peer connections between browsers. It handles the negotiation and management of media streams and their transport.
   
   
3. RTCDataChannel: Alongside audio and video streams, WebRTC offers a data channel allowing bidirectional arbitrary data transmission between peers.

Key WebRTC Architectures

Most WebRTC apps are based on P2P (peer to peer) architecture. While it is the safest way to enable low-ping high-security multimedia connections, it has certain disadvantages when it comes to handling large peer volume or different media streams. By intelligently using media server to act as a common converging point, developers can choose between (i) Multi-point Conferencing Unit (ii) Selective Forwarding Unit architecture to build WebRTC solutions.

Let’s understand all 3 WebRTC architecture types:

Peer 2 Peer

It adopts the concept of tin-can telephones where each user is interconnected with all the other participants. Any media is automatically passed to all the other recipients. Connections are established when each peers broadcasts their IP address. This is highly practical and commonly sought architecture that is simple and easy to set-up but it can be unstable at high peer volume and use up a lot of bandwidth.

Multi-point Conferencing Unit

The MCU architecture employ a central media server and all the peers connect to it. This enables multi-party communication and integrates various audio and video signals into a single stream. So every peer sends and receives the media directly to and from the other party with the media server in the middle and this allows MCU architecture to support larger number of connected peers. Additionally, MCU transmits standard signals that can be decoded on legacy systems as well without any decoder, making it compatible with most systems. However, this architecture is resource-heavy and it is costly to scale.

Wrapping Up

WebRTC has revolutionized real-time communication on the web, enabling developers to build powerful applications with seamless peer-to-peer connectivity. Its standardized APIs, along with media and data channels, simplify the process of integrating real-time communication capabilities into web applications. If you are looking for WebRTC development services or wondering what products are possible with WebRTC, get in touch with our experts at Vindaloo Softtech.