When COVID-19 effected the business around the world,  the demand for interactive live video platforms also growed.  The effective & efficient modle of deliver low-latency video with browser-based streaming is the need of most providers, now WebRTC is one option for developers and software engineers.

What Is WebRTC?

RTC means Real-Time Communications, WebRTC is an open framework for the web, and  enables through browsers.

This protocol creates a near simultaneous exchange of communication, utilizing a P2P connection between browser to browser, without any plug-ins.

WebRTC enables multiple browsers to communicate with one another directly, allowing for video conferencing between individuals and small groups. It also enables the development of high-quality RTC apps for browsers, allowing browsers communication through a set of common protocols.

When WebRTC works, it will employ three H5 APIs that are built into the Chrome, Firefox and Safari browsers. These APIs turn the browsers into encoders for video streaming, which then connect directly to other browsers for playback.

Is WebRTC Good for Your Business?

If you’re planning to use WebRTC in your business, you have to determine whether it’s the fit for your streaming workflow.

When to Use WebRTC? IT May be Good for You if:

You are requiring group video conferencing or real-time interactivity.

Depends on  the implementation, WebRTC is a good choose for group video conferencing and interactive, especially the cases with smaller audiences. As long as using WebRTC for your publish and playback, the technology supports real-time communication, in good network conditions, the latency is less than 0.5 seconds.

If you want to delay the playback time, or try to synchronize playback across multiple devices, you also can capture with WebRTC, but use HTTP Live Streaming for playback, using metadata and timecode to control the time you want referenced from playback.

You want to build a broadcast with a web app quickly and easily, without any heavy infrastructure. For WebRTC makes use of H5 APIs, it allows you to utilize many of the features built into the HTML5 programming language through a lightweight, embedded framework. This makes it easy for developers to build web-based apps using WebRTC, and for users to take advantage of the functional offerings without having to download clients or use plug-ins.

The features enabled by WebRTC include the ability to screenshare and to perform browser-based encoding, so users can live stream from their browser window to other viewers and participants on the web.

Benefits of Using WebRTC With Zego Streaming Engine

There are several benefits to building and deploying apps using WebRTC with Zego Streaming Engine as your media server. These include:

Browser-Based Encoded Streaming

Using only a web browser and Zego Streaming Engine, users can generate, record and deliver low-latency WebRTC-encoded streams—without the need for additional encoders or plug-ins.

Scaling WebRTC and Going Beyond P2P

Traditionally, WebRTC is known to be a nightmare when it comes to scalability. With Zego Streaming Engine, you can move beyond peer-to-peer communications and broadcast live streams to a large audience. The best way to scale your stream is to use the transcoder feature in Zego Streaming Engine to transcode the WebRTC stream into any standard output format.

Bandwidth-Optimized Video Conferencing

Zego Streaming Engine makes it easy to set up multi-person video conferences using WebRTC. It enables low-latency sessions between multiple users, while optimizing bandwidth by minimizing the number of connections each client must establish and maintain.

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