WebRTC is one of the most disruptive and transformative additions to real time communication and web technologies; its potential is huge and cannot be ignored. The ability to use just your browser – a tool you use daily – to conduct fully fledged web conferences, without the need for complicated multi-step procedures, or installations, is surely something to take notice of.

Otherwise known as Web Real-Time Communications, WebRTC is an open-source project – promoted by Google Chrome, Mozilla Firefox and others – it enables plugin-free Real Time Communications via Javascript API. It facilitates web browser to browser applications for voice calling, screen sharing, video chat, video conferencing and file sharing. The supported codec for WebRTC is currently VP8. WebRTC uses a server called Web Conferencing Server that in conjunction with a STUN Server  it is required to provide the initial page and synchronize the connections between two WebRTC endpoints.

WebRTC is supported by all modern browsers, including:

WebRTC support for these browsers is now built-in, an ideal solution that does not require any 3rd party components or plugins. The beauty of WebRTC is that when you are invited to a web meeting, no other action is needed other than clicking on a link that you’ve been provided and allowing relevant permissions. Installing or configuring yet another plugin might be “asking too much” for a lot of users.

WebRTC is gaining momentum and as browsers become embedded in smart TVs and IoT devices, it will be interesting to see who will be the next adopters.

When you initiate a call or video chat using WebRTC, it creates a direct, peer to peer connection between two devices using Stream Control Transmission Protocol (SCTP).

The process involves three key components: getUserMedia, which captures audio and video files, RTC Peer Connection, which manages the connection between two users, and RTC Data Channel, which transmits arbitrary data – anything from text to file sharing, making WebRTC versatile for various applications.

WebRTC is designed to be simple yet effective. You don’t need to worry about complicated setups or servers – everything happens in your web browser.

One of the most popular use cases is video conferencing. Platforms like Zoom and Google Meet rely on WebRTC to power real-time video communication, allowing teams to connect from anywhere without needing complex software installations.

Another common use is in voice calling within web-based applications. Think of customer service or sales teams using voice chat directly from a browser. WebRTC makes this possible without extra plugins.

It also supports file sharing and instant messaging, making it ideal for collaboration tools like Slack or Microsoft Teams, where users exchange files and messages instantly. Beyond that, WebRTC enables secure data sharing with its ability to handle arbitrary data, from real-time gaming data transfers to secure file transfers in a web environment.

WebRTC offers several key benefits that make it a preferred choice for businesses and developers. Let’s take a look at some of the standout advantages.

• Low latency: Direct peer to peer connections minimize delays, ensuring real-time communication is fast and efficient for video, voice, and data sharing.
• Scalability: Whether for small businesses or large enterprises, WebRTC easily scales to handle individual calls or large video conferences.
• Cost-effective: With no need for complex infrastructure or expensive licenses, WebRTC offers a budget-friendly solution for high-quality communication.

WebRTC is designed with safety in mind. It uses strong security protocols to ensure that the data being shared, whether audio, video, or arbitrary data, is protected from potential threats. Since WebRTC establishes direct peer to peer communication between users, it avoids sending data through central servers, reducing the chances of interception. This makes it a secure option for real-time communication, whether you’re making a video call, sharing files, or simply chatting online.

Datagram Transport Layer Security (DTLS) is used to secure data transmitted over WebRTC. It’s a requirement for all WebRTC solutions to be encrypted with Secure Real-time Transport (SRTP) Protocol. End-to-end encryption is also possible, if all recipients are on a SSL based connection.