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Live Streaming Using WebRTC – What is it?

Live Streaming Using WebRTC (1)

WebRTC (Web Real-Time Communications) is a standard that outlines the real-time transmission of streaming audio, video, and material from and to a browser without the need for plugins or other extensions. The standard enables you to convert your browser into a video conferencing terminal by just opening a web page.

While WebRTC video streaming was first sluggish to gain traction and acceptance, it has developed significantly in the recent decade. WebRTC has long been supported by Chrome, Firefox/Mozilla, and Opera, but Microsoft and Apple have lately joined the party, and the framework is now compatible with Safari, Microsoft Edge, Android, and iOS. WebRTC has been able to combine the advantages of an open-source, community-driven initiative with platform support as well as strong proprietary backing from a market leader.

The World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF) have approved it as an official standard, and it will continue to expand and include new use cases such as live feed processing and Internet of Things (IoT) integration.

What is WebRTC, and how does it work?

We’ll go over all there is to know about WebRTC technology for the ordinary user in this post. Let’s look at the project’s benefits and drawbacks, divulge some secrets, and explain how it works, as well as where and how WebRTC is utilized.

How WebRTC works On the client-side?

  • The user navigates to a page with an HTML5 element.
  • The user’s camera and microphone are requested by the browser.
  • To overcome NAT and Firewall, the JavaScript code on the user page controls the connection parameters (IP addresses and ports of the WebRTC server or other WebRTC clients).
  • The browser begins negotiating the audio and visual codecs used when it receives information about the interlocutor or the stream with the conference mixed on the server.
  • The encoding and streaming of data between WebRTC clients (in this example, the browser and the server) starts.

On the server-side

A video server is not necessary for data sharing between two participants, but it is essential if you wish to integrate several participants in a single conference.

The video server will accept media traffic from a variety of sources, transform it, and distribute it to WebRTC terminal users.

The WebRTC server will also accept media traffic from WebRTC peers and forward it to conference participants who are using desktop or mobile apps, if any are available.

WebRTC server

Benefits of the standard

  • No software installation required.
  • Very high communication quality thanks to:
  • Modern video (VP8, H.264) and audio codecs are used (Opus).
  • Automatic adjustment of stream quality to connection conditions.
  • Built-in echo and noise cancellation.
  • Automatic level control of participants’ microphones (AGC).
  • High degree of security: TLS and SRTP protocols are used to protect and encrypt all connections.
  • The desktop, for example, has a built-in system for collecting material.
  • Any control interface based on HTML5 and JavaScript may be implemented.
  • The ability to use WebSockets to integrate the interface with any back-end services.
  • You may include an open source project into your product or service.
  • True cross-platform: the same WebRTC application will run on any operating system, including desktop and mobile, as long as the browser supports WebRTC. When it comes to software development, this saves a lot of time and money.

Disadvantages of the standard

  • To organize group audio and video conferences, a video conferencing server is required that would mix video and audio from participants because the browser does not know how to synchronize multiple incoming streams with each other.
  • Because the standard only specifies ways for delivering video and sound, the implementation of techniques for addressing subscribers, monitoring their availability, exchanging messages and files, scheduling, and other features is left to the vendor.
  • In other words, you won’t be able to make a call from one developer’s WebRTC application to another developer’s WebRTC application.
  • Because group conference mixing requires a significant amount of processing power, this sort of video communication necessitates the purchase of a paid membership or an investment in infrastructure, with each conference requiring 1 physical core of a contemporary processor.

Find out how developing your own WebRTC can be useful by following the link: https://trembit.com/webrtc-development.html.