What is WebRTC?
WebRTC (Web Real-Time Communication) is a free, open-source technology that lets web browsers and mobile apps send audio, video, and data directly to each other without needing a middle server. It works right inside the browser, so users can talk or share files instantly.
Let's break it down
- Web: the internet pages you see in a browser.
- Real-Time: happening instantly, with almost no delay.
- Communication: sending and receiving information like voice, video, or files.
- Open-source: the code is free for anyone to use, modify, and share.
- Directly to each other: the data goes peer-to-peer, not through a central server (except for setup).
- Browser-based: no extra plugins or software downloads needed.
Why does it matter?
Because it makes live video calls, voice chats, and fast file sharing possible with just a web page, developers can build interactive experiences without costly server infrastructure, and users get smoother, more private connections.
Where is it used?
- Video-chat apps like Google Meet, Jitsi, and Zoom’s browser mode.
- Live customer-support chat windows that include screen sharing.
- Multiplayer browser games that need low-latency data exchange.
- Real-time collaboration tools (e.g., shared whiteboards, document editing) that stream video and cursor movements.
Good things about it
- No plugins required - works in most modern browsers out of the box.
- Low latency, ideal for real-time voice and video.
- Peer-to-peer data reduces server bandwidth costs.
- Built-in security with encryption (DTLS/SRTP).
- Open standards mean wide community support and continuous improvement.
Not-so-good things
- Requires a stable internet connection; poor networks cause glitches.
- NAT and firewall traversal can be tricky, sometimes needing extra servers (STUN/TURN).
- Browser compatibility varies slightly, especially for advanced features.
- Limited control over quality and codecs compared to native apps.