WebRTC (Web Real-Time Communication) is an open-source technology that enables real-time audio, video, and data transmission directly between browsers and devices without requiring additional plugins or software. It is widely used in video conferencing, live streaming, online gaming, and peer-to-peer (P2P) file sharing. WebRTC is developed by Google and standardized by W3C (World Wide Web Consortium) and IETF (Internet Engineering Task Force). It is supported by major browsers like Chrome, Firefox, Edge, and Safari, making it a key technology for real-time communication on the web.
WebRTC establishes direct peer-to-peer (P2P) connections using several key components:
Signaling: Exchanges metadata (e.g., IP addresses, session details) via external protocols like WebSockets or SIP to initiate connections.
STUN/TURN Servers: Helps bypass firewalls and NAT restrictions, ensuring connectivity.
Peer Connection: Uses Secure Real-Time Transport Protocol (SRTP) to encrypt media streams and DataChannel API for P2P data exchange.
Once established, WebRTC allows low-latency, encrypted, and high-quality media transmission between users.
Low Latency: Supports sub-second communication, making it ideal for live interactions.
End-to-End Encryption: Uses DTLS-SRTP for secure media and data transmission.
Cross-Platform Compatibility: Works across web, mobile, and embedded systems.
Adaptive Network Handling: Adjusts media quality dynamically based on network conditions.
No Additional Software Required: Runs natively in modern browsers.
| Feature | WebRTC | HLS | DASH | RTMP |
|---|---|---|---|---|
| Latency | Sub-second (<500ms) | 5-30s | 3-10s | 2-5s |
| Transport Protocol | UDP (SRTP, SCTP, DTLS) | HTTP (TCP-based) | HTTP (TCP-based) | TCP (RTMP) |
| Ideal Use Cases | Video calls, gaming, live chat | OTT, VOD, live streaming | OTT, VOD, adaptive streaming | Legacy live streaming |
| P2P Support | Yes | No | No | No |
| Encryption | Yes | No (unless DRM-encrypted) | No (unless DRM-encrypted) | No |
Scalability: P2P connections require significant bandwidth in multi-user scenarios.
Firewall & NAT Issues: May require STUN/TURN servers for connectivity.
Limited CDN Support: Unlike HLS/DASH, WebRTC is not optimized for traditional CDN delivery.
To enhance scalability, many platforms integrate WebRTC with Selective Forwarding Units (SFU) or Multipoint Control Units (MCU) to handle large-scale real-time communications.
WebRTC is a critical technology for real-time, interactive communication, offering low-latency, secure, and browser-native capabilities. While it excels in applications like video conferencing, gaming, and live interactions, large-scale deployments require hybrid models with media servers or CDNs for better efficiency.