The magic of real-time communication

Real-time communication (RTC) is a key technology that allows the transmission of audio, video, and data with minimal delay, high reliability, and maximum availability. Chroom, using advanced decentralized network algorithms, transforms this technology by creating an innovative way to organize interactions over the internet.

What is WebRTC and How It Works in Chroom?

WebRTC (Web Real-Time Communication) is a set of standards that allows the transmission of audio, video, and data directly between browsers or mobile devices. What sets WebRTC apart from traditional data transmission technologies is that it doesn't require additional plugins or software — everything works directly in the browser. WebRTC is widely used for video calls, online conferences, and real-time file sharing.

How Does WebRTC Work in Chroom?

WebRTC includes three key components:

1. getUserMedia — allows access to the device’s camera and microphone for audio and video transmission.

2. RTCPeerConnection — establishes a direct connection between devices to transmit data in real time.

3. RTCDataChannel — used for exchanging data such as text, files, and other content.

In Chroom, these components work on a decentralized basis, allowing for improved performance by:

• Enhanced security, as data is not transmitted through central servers.

• High reliability, as the network architecture is protected from single points of failure.

• Ultra-low latency, since data is transmitted directly between participants via optimized routes.