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WebRTC vs WebSocket: Real-Time Communication Patterns

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Real-time: two different problems, two different solutions

"Real-time" covers very different requirements. Chat and collaborative editing need reliable ordered delivery — a message must arrive exactly once, in order. Audio/video calls need low latency — a 300ms delay makes conversation unnatural, but a slightly blurry frame is fine. A lost audio packet is better dropped and concealed than waited for.

These requirements are fundamentally in tension: TCP's reliability mechanisms (retransmission, ordering) add latency. This is why WebSocket (TCP-based, reliable) and WebRTC (UDP-based, lossy-tolerant) exist as separate technologies for different use cases.

Packets in motion — click to inspect:

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