Skip to main content

WebRTC (Web Real-Time Communication)

WebRTC is an open-source protocol that enables real-time audio, video, and data communication directly between browsers and devices. Its primary purpose is to allow developers to build live communication features, such as video conferencing, file sharing, and live streaming, without the need for additional software plugins or downloads. WebRTC works by establishing a peer-to-peer connection between two or more devices, allowing for low-latency, secure, and high-quality communication that is the backbone of modern web-based collaboration tools.

Example: A fintech company is developing a new, highly secure login feature that uses a WebRTC-based video call for real-time identity verification. The quality assurance team would test this feature to ensure a seamless and secure user experience. They would:

  1. Test the Peer-to-Peer Connection: Verify that the user's browser can establish a direct, real-time video stream to the company's verification agent.
  2. Evaluate Performance: Simulate a user on a slow network to see if the video quality gracefully degrades without freezing or dropping the call.
  3. Confirm Security: Check that the video and audio streams are encrypted to prevent a third party from intercepting the communication.

This process ensures that the WebRTC login is reliable, secure, and works for all users.