Most of you are probably reading those words from the comfort of your own home, even though it’s the middle of the week, and all of us busy bees are at work. The world of office work is not the same anymore, but one thing never changes – and it’s the meetings. We all need to put our time into business discussions, long brainstorming sessions over new projects, or just attend one more important training. With the power of a decent webcam with a headset, we’re all set and ready to jump in.
After the work is over, your mother-in-law renovated the kitchen and wants to show you around – online. Your college friends are cracking a cold one open while watching a show together – online. You want to join a French language course and prepare for your trip to Paris this summer – and the course is online too.
A few years ago, spending this much time in online calls would probably send your head spinning – horrible, pixelated video, low-res audio that constantly breaks up, constant lag. Spending more than 15 minutes in a video chat was not for the faint of heart.
Crisis breeds innovation. Over the past few years, we got a crash course in being in video chats all the time. And old VoIP apps were not up to snuff for prolonged use and heavy volumes. Luckily, a little thing called WebRTC protocol was gaining traction, and the time to really put it to the test was upon us.
WebRTC is free to use and completely open-source, supporting peer-to-peer (P2P) audio, video, and data transfer. It can facilitate real-time communication (RTC) capabilities in browsers and standalone applications with its API. The widespread availability of WebRTC protocol on all major platforms makes it easy for developers to implement it in their voice and video communication solutions.
WebRTC applications don’t require any third-party plugins or apps to work. They can be simply embedded directly into any website – WebRTC protocol runs natively on Google Chrome, Mozilla Firefox, Microsoft Edge, Safari, Opera, and more popular browsers. And thanks to its peer-to-peer capabilities, the most simplistic WebRTC-based solutions can run independently without needing a dedicated host server.
WebRTC is powering many popular communication apps – for example, you can see it in action when using Discord, Zoom, Google Meet, or Facebook Messenger.
WebRTC works by facilitating the direct path between two or more clients and ensuring the secure transfer of data and audio or video streams simultaneously in all directions. The connection is established through a signaling device that negotiates the paths and provides the correct data routing. The signaling server can either be public to create quick P2P connections or a dedicated, private server for handling more enormous volumes of both data and users, as well as additional layers of security since the servers can require additional authentication and have limited accessibility (or just be completely private).
Many developers prefer implementing open-source projects into their work because of the great flexibility, transparency, and ease of use. That’s why WebRTC is an optimal solution for many. With its broad compatibility, it’s easily accessible to almost all users, except ones relying on legacy browsers, e.g., Internet Explorer. The communication is done almost precisely in real time, as P2P connectivity reduces the latency associated with long server jumps and heavy transcoding of communicators of yesteryear.
With bigger volumes come bigger resource needs – the more users you want to have on your WebRTC-based solution, the more it’s going to take to keep the connections optimized and fast. P2P connections are only beneficial when serving a few communication routes simultaneously. But even when using dedicated servers to operate your application with dozens or more people on each end, the speed increase with WebRTC is still going to be very noticeable compared to classic protocols like VoIP (Voice over Internet Protocol). Are you going to start your WebRTC project? Here are 3 things to consider before you start your WebRTC project.
We all need real-time communication right now.
Remote work, online gaming, quality time with your cousin across the ocean – all of those need a stable, reliable, high-quality online connection to communicate effectively. With challenges arising in recent times, creating simple solutions that bring people closer is essential. When building applications that strive to enhance human connections over the internet, with WebRTC, what you’re getting is a powerful tool that’s actually very easy to implement and use while requiring way fewer resources – bridging the gap between what’s doable for developers and what just works for the end-user.
Running a WebRTC app through a private server has immense benefits. Secure your communications and allow for substantial data volumes in your communication solution by using our dedicated servers, finetuned for optimal signaling and routing. Combined with our ultra-low latency network, it’s no wonder it handles the vast majority of traffic on such popular apps like Discord without even breaking a sweat.