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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 and How It Works

The world of web-based communication is changing at a speed you might not even imagine.  Now, for making a video conferencing call, sharing a file—all that you need is a URL, thanks to the power of WebRTC for opening the door of infinite possibilities for developers and businesses.     

Developed by Google in 2011 as an open-source framework comprising standards, protocols, and JavaScript APIs, WebRTC or Web Real-Time Communications facilitates the peer-to-peer transfer of audio, video, and data without plugins, native applications, or any third-party proprietary software.  

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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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WebRTC Use Cases, Challenges, and Trends

What is WebRTC?

An open-source project released by Google in 2011, WebRTC provides API-based communication between web browsers and mobile applications, including transmissions of audio, video, and data. Eliminating the need for native plugins and app installations makes these connections user-friendly and supported by all the major browsers and mobile operating systems.

The adoption of WebRTC in the tech community has grown dramatically in the past few years. Facebook, Amazon, and Google are among the significant technology companies that implemented WebRTC to make their web applications faster, reliable, and more secure.

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What Is Transcoding? Why Is Transcoding Important for Streaming?

If you are in the streaming industry or have some interest, you may have heard the word transcoding a lot. You may know a little about what transcoding is, or you may be wondering what this word you hear all the time. However, regardless of your situation, after this blog post, you will have full control of the transcoding process. Let’s not forget that the video transcoding process is very important for live broadcasts, especially if you want to use adaptive streaming in live streams to reach more audiences. We will find an answer to the question of what is transcoding.

But… To really understand what transcoding is and how it affects the success of your streams, firstly, we need to talk about encoding and need to understand what encoding is. We cannot talk about transcoding without talking about encoding.

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Predictable Pitfalls of Scaling VoIP to Cluster

Avoid these pitfalls of scaling VoIP to cluster.

VoIP technologies have a reputation for being rather complex and not without good reason. This is not to say, however, that any mid-level developer will have much difficulty finding and handling one of many available open-source servers. The only problem is the lack of information and real-world examples, as VoIP server homepages rarely have configurations that go beyond the very simplest option, and even when they do, they may be not up-to-date.

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