Real-time Communications with WebRTC

Quarantines and working from home have made remote communication a necessity,

enabling billions of users to adapt to video calling. One of the video communication

applications that has increased in use and popularity with the Covid-19 pandemic is WebRTC. WebRTC is a completely free HTML5 specification. You can add real-time media

communications with WebRTC directly between the browser and devices, enabling audio and video communication to work on web pages. Moreover, it does this without the need to

install a browser add-on.

WebRTC is embedded in browsers as an open-source project, but you can pick it up and

adapt it for your own needs. WebRTC is available today in all modern browsers such as

Google Chrome, Mozilla Firefox, Apple Safari, Microsoft Edge. It is estimated that more than

one billion minutes and 500 terabytes of data are transferred each week, with WebRTC traffic

originating from browser communications alone. You can also take it and integrate it into an

app or embedded device without the need for a browser. You also get a vibrant and dynamic

ecosystem of commercial offerings from companies that help you build your products around WebRTC.

So how does WebRTC work?

WebRTC is an open-source and completely free project available in all modern browsers - for

commercial or private use. With the fact that it is a constantly developing technology, it gives

you the confidence of having a technology that will work for you for many years. This

advantage paves the way for the creation of new use cases and business models. Many

events such as large-scale webinars and live broadcasts are possible with WebRTC

technology.

But WebRTC is not limited to browsers only. It is also available for mobile applications thanks

to its portable source code and is already used in many mobile applications. It is possible to

run WebRTC anywhere with SDKs that will enable you to use it in both mobile and embedded

environments.

Previously it was often necessary to use C/C++ to create anything that allowed a voice or video calling. This resulted in longer development times and higher development costs. WebRTC, on the other hand, comes with a Javascript API layer on top that you can use inside the browser by replacing the C/C++ infrastructure with a Javascript API. This makes it much easier to develop and integrate real-time communications with WebRTC anywhere.

Internally, WebRTC is still mostly implemented using C/C++. However, most developers using

WebRTC will not need to delve deep into these layers to develop their applications.

With WebRTC, you can access your device's microphone, the camera on your phone or

laptop, or a screen. You can capture the image of the user and then have this screen shared

or recorded remotely. It enables live interactions, in real-time. WebRTC is not limited to audio

and video. It allows sending any arbitrary data.

WebRTC is quite powerful and versatile beyond voice or video calling. You can use it for

group search, adding records, or just data distribution. WebRTC takes the concept of a

communications service and reduces it to a feature within a different service type. Now you

can take it and add communications in the business processes you need within your application or business.