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The Most Comprehensive Guide on WebRTC

WebRTC or Web Real-Time Communications, though a relatively new web technology, has taken web-based communication at an entirely new level with the promise of heralding into a brave new world of communication on the horizon. The free, open-source WebRTC project makes use of a set of JavaScript APIs to facilitate peer-to-peer communication between web browsers and different devices. The question remains what makes it so popular.

A big draw with WebRTC is it eliminates the use of plugins or third-party software to facilitate real-time communication, helping achieve the ultimate goal of moving in a plugin-free world.

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What Is WebRTC?

WebRTC stands for web real-time communications. It is a very exciting, powerful, and highly disruptive cutting-edge technology and standard. WebRTC leverages a set of plugin-free APIs that can be used in both desktop and mobile browsers and is progressively becoming supported by all major modern browser vendors. Previously, external plugins were required in order to achieve similar functionality as is offered by WebRTC.

WebRTC leverages multiple standards and protocols, most of which will be discussed in this article. These include data streams, STUN/TURN servers, signaling, JSEP, ICE, SIP, SDP, NAT, UDP/TCP, network sockets, and more.

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Ultra Low Latency Video Streaming and 7 Use Cases

Especially with COVID-19, people started to demand more live streams. But especially in some live streams which we will talk about in the rest of the blog post should be really 'live' to satisfy the audience. Let’s start with how popular live streaming is. According to the Cisco Annual Internet Report; Increasing internet connection speeds will enable a higher resolution to live video streaming and live video streaming will become popular in every field.

With the popularity of live streaming, the need for ultra low latency video streaming is increasing day by day.