WebRTC: Real-Time Browser-Based Communication

WebRTC (Web Real-Time Communication) is an open-standard technology that enables real-time audio, video, and data exchange directly between browsers or mobile apps — without requiring additional plugins or intermediary media servers. It is widely used in applications such as video calls, online collaboration tools, telehealth, gaming, and customer support platforms.

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Core Characteristics

• Real-Time Media Streaming
  WebRTC enables low-latency video and audio communication — ideal for conferencing, gaming, and live collaboration.
  
  
• Peer-to-Peer (P2P) Communication
  Data is exchanged directly between devices, minimizing latency and reducing infrastructure costs. When direct connection is blocked by networks, WebRTC uses TURN/STUN servers for fallback.
  
  
• Cross-Platform Support
  WebRTC runs natively in major browsers (Chrome, Firefox, Safari, Edge) and mobile environments, enabling unified communication experiences.
  
  
• Secure-By-Design
  All media and data channels are encrypted using DTLS and SRTP, ensuring secure transmission.
  
  

Key Components

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Example Use Case

A telemedicine app uses WebRTC to:
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• Capture video from the patient and doctor via getUserMedia()
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• Establish a secure P2P channel using RTCPeerConnection
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• Exchange chat messages and files through RTCDataChannel
  
  

Result: real-time encrypted consultation without installing any additional software.

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Related Terms

• WebSockets
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• Single Page Applications (SPA)
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• Text-to-Image