Use cases of WebRTC, how it works and benefits of the technology explained in a nutshell.
What is WebRTC?
WebRTC is an HTML5 specification that you can use to add real time media communications directly between browser and devices.
WebRTC enables for voices and video communication to work inside web pages.
And you can do that without the need of any prerequisite of plugins to be installed in the browser.
It was announced in 2011 and since then it has steadily grown in popularity and adoption.
By 2016 there has been an estimate from 2 billion browsers installed that are enabled to work with WebRTC. From traffic perspective, it has seen an estimate of over a billion minutes and 500 terabytes of data transmitted every week from browser communications alone.
WebRTC has increased in popularity and use throughout the COVID-19 pandemic. Quarantines and work from home made remote communications a necessity, indoctrinating billions of users about the use of video calling. The end result has been a surge in the use of WebRTC:
In 2021 WebRTC got officially standardized, removing all doubts about its future prospects. Today, WebRTC is widely popular for video calling but it is capable of so much more.
A few things worth mentioning:
- WebRTC is completely free
- It comes as open source project that has been embedded in browsers but you can take and adopt it for your own needs
- This in turn has created a vibrant and dynamic ecosystem around WebRTC of a variety of open source projects and frameworks as well as commercial offerings from companies that help you to build your products
- WebRTC constantly evolving and improving, so you need to keep an eye on it (e.g. see hiring WebRTC developers)
- See also: The state of WebRTC open source projects
Covered in this video:
- What is WebRTC?
- Current state of adoption
- Why is it so much more than just a video chat enabler
- The power of “Open Source”
- How WebRTC works
- Five reasons to choose it
(this article was updated in October 2022)
WebRTC stands for Web Real-Time Communications.
Web is simple – it means that what we are doing works “over the web” and inside a browser. The browser part means that all modern browsers support WebRTC. If you run this inside a native application I will still be considering it as WebRTC. To me, it is the thought that counts, or more accurately, the implementation of WebRTC (or parts of it) are quite popular as a starting point in native applications. This is due to the quality of the WebRTC media engine (as implemented by Google) AND due to the fact that it makes it easier to communicate across native applications and web applications this way.
RTC, or Real-Time Communications means that whatever WebRTC does – it does in real time. Its focus is on sending the data it has as fast as possible – making sure to use low latency techniques to get things done. Whenever possible.
If we’re moving away a bit from the word-meaning of WebRTC, then this is the definition I usually use to define WebRTC:
Lets break it down a bit:
- WebRTC offers real time communication natively from a web browser
- WebRTC is part of the web browser. Every modern web browser today implements WebRTC
- It offers the ability to create real time communication applications and experiences
- WebRTC is a media engine. There were other media engines before WebRTC and there will probably be others after it. In that sense, there’s no “innovation” here
So, how does WebRTC work?
Code and API
It is important to understand from where we are coming from: If you wanted to build anything that allowed for voice or video calling a few years ago, you were most probably used C/C++ for that. This means long development cycles and higher development costs.
WebRTC today is available in all modern browsers. Google Chrome, Mozilla Firefox, Apple Safari and Microsoft Edge support it.
You can also “take” it and integrate it into an application or an embedded device without the need of browser at all.
Media and access
What WebRTC does is allow the access to devices. You can access the microphone of your device, the camera that you have on your phone or laptop – or it can be a screen itself. You can capture the display of the user and then have that screen shared or recorded remotely.
Whatever it does is in real time, enabling live interactions.
WebRTC isn’t limited to voice and video. It allows sending any type of arbitrary data.
There are several reasons WebRTC is a great choice for real time communications
- First of all, WebRTC is an open source project
- It is completely free for commercial or private use, so why not use it?
- Since it is constantly evolving and improving, you are banking on a technology that would service you for years to come
- WebRTC is a pretty solid choice – It already created a vibrant ecosystem around it of different vendors and companies that can assist you with your application
- It is available in all modern browsers
- This has enabled and empowered the creation of new use cases and business models
- From taking a Guitar or a Yoga lesson – to cloud gaming and social networking – to medical clowns or group therapy – to hosting large scale professional Webinars and live broadcasts; WebRTC is capable of serving all of them and more
- WebRTC is not limited to only browsers because it is also available for mobile applications
- The source code is portable and has been used already in a lot of mobile apps
- SDKs are available for both mobile and embedded environments so you can use WebRTC to run anywhere
- WebRTC is not only about for voice or video calling
- It is quite powerful and versatile
- You can use it to build a group calling service, add recording to it or use it only for data delivery
- It is up to you to decide what to do with WebRTC
- WebRTC takes the notion of a communication service and downgrades it into a feature inside a different type of service. So now you can take it and simply add communication in business processes you need within your application or business
What is WebRTC used for?
You can group WebRTC applications into 4 broad categories:
- Conversational voice and video – the obvious one. Applications that need the ability to have a person communicate with others in real time and in a conversational manner. These will more often than not end up using WebRTC
- Live streaming – while WebRTC isn’t the most popular choice for streaming, it is one of the best technologies available for low-latency live streaming. If you need to stream something to one or more users and maintain really low latency to enhance the interactivity (things like cloud gaming, gambling, auctions, webinars, etc) – then WebRTC might be a great choice
- Data transfer – you can send voice and video with WebRTC, but you can also send arbitrary data. This can be used to share huge files between machines with little need for server space for example. Or it can be used to create a bittorrent like experience
- Privacy – since WebRTC runs direct between browsers, it is sometimes used to enhance privacy. Doing this by simply not sending the media or data via servers at all
Overview of Use-Cases with WebRTC
The use cases where WebRTC comes in handy seem endless. Every so often, I hear of a new way that WebRTC is being used to solve yet another problem.
Here are some of the main use cases you’ll find for WebRTC out there:
- Unified communications – voice and video calling, 1:1 or group sessions
- Contact center communications – client/agent, visual assistance, remote assistance, etc
- Watch parties – watch television or a sports event together
- eCommerce and retail – from one-to-one high touch sales to live broadcast for sales events and promotions
- Telehealth, online education, legal proceedings, remote travel, fitness, dancing, tutoring, coaching, … – conduct remotely and virtually verticalized sessions you would have done in-person in the past
- Teleoperations – drive cars, forklifts, trucks, drones, boats, submarines, … – remotely
- Virtual and hybrid events – conduct webinars, large meetings and events online
- Low latency broadcasting – broadcast a sports game, auction or interactive sessions to a large audience at sub second latency
- Cloud gaming – render the visuals of a game in the cloud and send it in realtime to the player
- Machine remoting – operate a remote machine (high performance machines or highly secured/configured machines) as if it was a local one
- Virtual spaces and the metaverse – meet people in a synthetically rendered virtual environment in 2D or 3D
So what other choice do you really have besides using it?
The idea around WebRTC and what you can use it for are limitless. So go on – start building whatever you need!