WebRTC

WebRTC (Web Real-Time Communications) is an open-source technology that allows the transmission of audio, video, and other data in real-time directly between browsers or other supporting applications without the need for additional plugins or software.

Simply put, WebRTC turns your browser into a full-fledged tool for video calls, voice communication, file sharing, and even online gaming without needing to download Skype, Zoom, or other software.

Where is WebRTC Used?

• Video Conferencing and Calls: Google Meet, Discord, and Zoom (partially) use WebRTC.
• Streaming Services: For example, for live broadcasts directly from the browser.
• Remote Desktop and Support: To share your screen or access another one.
• File Sharing: For example, Peer5 for distributing large files directly between users.
• Online Gaming: For multiplayer mode with minimal latency.
• Adding AR/VR to the Browser: For interacting with the camera in real time.

Frequently Asked Questions

What are codecs in WebRTC?
These are compression and decompression algorithms for audio and video streams. They are needed to reduce the amount of data transmitted over the network without a critical loss of quality.

Which video codecs are most commonly used in WebRTC?
VP8, VP9, H.264, and the increasingly popular AV1. Browsers automatically negotiate the most suitable codec supported by both participants.

Which audio codec is best for WebRTC?
Opus is considered the modern de facto standard due to its excellent sound quality even at low bitrates and high resilience to packet loss.

Why doesn’t my browser support a certain codec?
Codec support depends on the browser and operating system. Our tester will show you which codecs are available in your current browser.