What is WebRTC?
WebRTC (Real-Time Communication) is a free and open-source project that allows you to add real-time voice and video communication capabilities to your applications. With WebRTC, it’s easy to build a real-time communication application system as it allows web apps and sites to stream audio and video media without the need for a third-party system.
WebRTC consists of a set of APIs (Application Programming Interfaces) and protocols that combine to set standards that enable data sharing and peer-to-peer connectivity.
While WebRTC is not an application itself, a lot of applications have used it, some of which include SignalWire, Google Meet, Facebook Messenger, Discord, Amazon Chime, Houseparty, GoToMeeting, Peer5, WhatsApp, and Snapchat. Although many of these apps do not adopt WebRTC 100%, we can expect that they will do so as time goes by and more updates are added to the system.
A Brief History of WebRTC
After the release of Google Chrome, it was discovered that the web system had limitations in real-time communications. There was no generally accepted and standardized system across all browsers that permits data transfer directly between two people. Even worse, there was no default implementation of such a system in any browser at all.
In May 2011, Google embarked on a project to build a system that allows seamless data transfer on a mutual platform without the need for plug-ins or third-party apps. Mozilla, Opera, Microsoft, and Apple expressed interest in the project after a few years and joined Google in searching for answers. This project is now known as WebRTC.
In January 2011, Ericsson Labs built the first implementation of WebRTC with a modified WebKit library. The W3C made its first sample for the spec in October of that same year. As of 2014, Google Hangouts was somewhat using WebRTC. The WebRTC 1.0 spec progressed into Candidate Recommendation from the Working Draft in November 2017. As of January 2021, the WebRTC 1.0 spec has further moved to Recommendation.
With the advent of FaceTime, Skype, Google Hangout, and the like, it became easier to communicate directly. They are beautiful improvements to technology, however, they still fall short of the standards set by the WebRTC project. The reason is simple: recall that the WebRTC project aims to allow real-time communications without the need for plug-ins or third-party apps.
FaceTime and Skype require that both parties at each end of the video call use a mutual technology. For instance, you cannot FaceTime anyone who doesn’t use an Apple device. You would have to use another platform. Another limitation is you have to be prepared ahead of the call to ensure that the app downloads correctly or plug-ins are in perfect condition. It is only after these conditions are met that you can join a video conference.
This is where and how it all started. In webrtc.org, the original website of the project, it is stated clearly that the aim of the project is “to enable rich, high-quality RTC applications to be developed for browsers, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.”
What Problems does WebRTC solve?
While phone calls, emails, text messages, and other forms of communication are still valuable today, real-time communication is becoming more important. The need for genuinely immersive conversations between people makes real-time communication very important in our daily lives. Think you’re already harnessing all there is to communications? Think again - there is more to be seen.
There are still limitations present in communication that WebRTC aims to overhaul and as time goes on, there will be more improvements to the system. We will enjoy a better communications experience than any other time in history. We are enjoying some already.
Let us dive into some existing problems that the emergence of WebRTC has succeeded in surmounting.
High cost of telecommunication: There is no need to be charged per minute or call. All you need is an internet connection in order to experience the beauty of WebRTC. Unlike the traditional communications systems that mostly have to do with voice conversations, WebRTC provides you with the ability to have a premium video experience and communicate with a person as if s/he were standing next to you. WebRTC is a solid rival to the status quo.
Platform and device incompatibility: A real-time voice or video communication can be made without the need for a mutual operating system. You can use a WebRTC-enabled device to connect to another WebRTC-enabled device or WebRTC media server regardless of the possible differences in operating systems and web services applications. Standard APIs from the W3C (World Wide Web Consortium) and protocols from the IETF (International Engineering Task Force) collaborate to make this possible.
Security concerns and privacy breaches: WebRTC has an end-to-end voice and video encryption that is always activated. Encryption and authentication of data (voice and video) are made possible using.” Secure RTP protocol, which is particularly helpful over Wi-Fi networks. This stops the loss of data, manipulation of data, and leaking of data to third parties. In simple terms, no one outside both parties involved in a call can listen to the conversation, let alone record it.
Poor voice and video quality: WebRTC provides high-quality audio that has never been experienced before. The Opus codec and VP8 codec are the technology used for WebRTC’s audio and video, respectively. These technologies allow the audio and video systems to function uniformly and steer clear of harmful codes resulting from codec downloads.
Session establishment limitations: WebRTC solves the problem of unreliability by avoiding media transferred by servers. This is seen in NAT (Network Address Translators), which impedes and can block other communications and collaboration protocols. This decreases dormancy and server load and thus increases quality.
Limited media streams: WebRTC adapts to changes in network conditions. It is an adjustive system that responds to bandwidth availability, adjusts communications quality, discovers, and stops excess network load. The RTP (Real-Time Transport Protocol) Control Protocol and Secure Audio Video Profile with feedback make adaptability possible in WebRTC.
The sending browser interacts with the receiving browser by collecting information from it and analyzing the data in response to changes in network conditions. WebRTC also allows the negotiation of several media types and endpoints, whether in size or format. This produces an efficient use of bandwidth, which in turn makes premium voice and video communication possible.
Constant updates issues: WebRTC supports interoperability between existing voice and video systems, which includes devices utilizing Jingle, SIP (Session Initiation Protocol), XMPP (Extensible Messaging and Presence Protocol), and the PSTN (Public Switched Telephone Network). VoIP applications need to be regularly updated.
If one update is missed, it could lead to the introduction of malicious systems and third-party interruption. Most software does not update automatically, and we humans ignore it. We skip it because it consumes our time and data. WebRTC helps to combat this problem by eliminating the need to deploy the client software.
Slow application development process: WebRTC solves the problem of tardy app development for developers. In earlier times, there was no streamlined development process for app development, so application implementation took time. Standardized APIs make app development possible without the need for detailed knowledge of WebRTC.
WebRTC is an open-source API that is free and present in all available browsers. It is so simple to use, all you have to do is click a link, and you join a video meeting. You can also adopt and take WebRTC for your own needs and use it for commercial purposes – to build products for companies. WebRTC is not a stagnant technology. It changes regularly and keeps getting better, so you have to pay close attention to stay ahead.
In summary, WebRTC solves the problem of growing expectations of user experience. Experience means a lot to people, and everyone is looking forward to having the best in their search to stay connected with the world. WebRTC provides one of the simplest ways to access VoIP and voice connectivity. WebRTC allows you to communicate live with people as if they were sitting right next to you at the bus stop.
Additionally, with the rise of 5G in today’s world, people with WebRTC-supported mobile devices will have access to HD video content from their phones anywhere in the world. High-definition video conferencing will be possible for people with WebRTC-supported devices. Combining 5G and WebRTC will bring tremendous improvements to the way we perceive our virtual environment.
Every Browser does not have the same WebRTC features at the same time. Different browsers may have some features non-existent in other browsers, which explains why some WebRTC features work in some browsers and not in others.
Google Chrome, Mozilla Firefox, and Opera all support WebRTC on both desktop and Android. Other browsers include:
- Microsoft Edge
- Brave Browser
- Chrome OS
- Firefox OS
- iOS (MobileSafari/Webkit)
- BlackBerry 10
- Tizen 3.0
Furthermore, WebRTC support for these browsers is now inbuilt. It is a perfect solution as it does not depend on any third-party component or plugin to function.
As a user, all you have to do is click the provided link and grant relevant authorization, and you are connected. With the rise of Smart TVs and IoT devices with in-built browsers, it will be exciting to see how the manufacturers can adopt WebRTC to improve communication.
Current adoption of WebRTC
Research has shown that WebRTC has been growing gradually since 2019 at 6% annually and will continue in the same manner until 2025. By 2025, WebRTC will be valued at over $21 million, a significant rise from its 2018 value of $1.6 million.
The global COVID-19 pandemic in 2020, which hit the planet and caused a global lockdown that rendered many businesses inactive, contributed significantly to the adoption of WebRTC as more communication applications needed to be built in a short amount of time.
People still needed to make money while staying at home. Essential products still had to be sold to people for survival. Companies had to employ tech-savvy people. This led to the rapid adoption of WebRTC for communication between people around the globe.
After the lockdown was lifted, it became increasingly difficult for businesses to go back to their regular routine. Remote working was added to the normal business routine. It is interesting to note that over 11 million meetings are held daily by American businesses, many of those now taking place online. Additionally, over 30% of customers today hope that companies engage them in a live conversation via their website.
Times are changing fast, and in 2021, it has become increasingly important that visual communication between people exists to ensure the smooth running of businesses. We are no longer interested in just hearing voices; we want to see faces too. We want to see the reactions of the people we deal with on a daily basis.
Today, organizations around the globe are utilizing WebRTC technology to conduct interviews for prospective candidates, set up training for workers and newbies, plan strategies, and even hold meetings that are simply for social interaction and leisure. With the rise in freelancing and remote jobs, WebRTC is gradually replacing face-to-face meetings and even human interactions outside workplaces.
Health care and defense institutions use WebRTC for training, cloud gaming and social networks have adopted live streaming and interactive broadcasts, and the entertainment industry is trying to find a remote solution to enable the audience to be present in the studios. Additionally, experts are trying to utilize WebRTC to the fullest and recreate the in-stadium experience in sports. Even family and friends use WebRTC products in their day-to-day activities.
What are the alternatives to WebRTC?
WebRTC is the only browser system that aids voice and video communication. However, other similar real-time communication products are mostly powered by VoIP technologies to support mobile and PC apps. Here are a few of them:
Element: It is a messaging app based on the Matrix protocol. It is secured with end-to-end encryption. It is decentralized and boasts of premium control, easy connections, and interoperability. It supports Mac.
Autobahn: This project utilizes The WebSocket Protocol and The WAMP (Web Application Messaging Protocol) network protocols to provide open-source VoIP communication. It supports macOS and Windows.
Dust: It is a free and proprietary project. You have complete control over your digital affairs with this system. It supports Android and iPhone.
Librem Chat: It is a free and open-source system that is secured with end-to-end encryption. Millions around the world use it to share files and make real-time communication. It supports Android and iPhone.
Gevent: It is an open-source coroutine-based project. It is a Python networking library that provides a high-level simultaneous API using greenlet. It supports Mac and Windows.
For streaming, there are also other alternatives to WebRTC like Flash, HLS, and MPEG-DASH.
Although the alternatives listed above are decent, they cannot be compared to WebRTC in terms of communication experience. WebRTC has undergone several improvements over the years, and it edges them by a distance. WebRTC remains the best choice for premium real-time communication.