WebRTC vs WebSockets: What Is the Difference?

WebRTC and WebSockets are both real-time technologies, but they solve fundamentally different problems. WebSockets connect a client to a server for bidirectional messaging. WebRTC connects two clients directly for audio/video/data. Here is when to use each — and how they work together.

The short answer

WebSockets are for client-to-server real-time messaging (chat, notifications, live data). WebRTC is for peer-to-peer real-time media (audio, video, large data). They are complementary, not competing — most real-time apps use both.

What WebSockets do

A WebSocket is a persistent, bidirectional TCP connection between a browser and a server. The server can push messages to the client at any time (no polling needed). WebSockets are ideal for chat, live notifications, real-time dashboards, and any scenario where the server needs to push data to the client.

WebSockets carry text and binary data, but not audio/video media. All traffic goes through the server. Latency is the round-trip time to the server (typically 20–100ms depending on geography).

What WebRTC does

WebRTC is a peer-to-peer protocol for real-time audio, video, and arbitrary data. Once two peers establish a connection (using a signaling server — see above), media flows directly between them without passing through any server. This gives the lowest possible latency for audio/video because there is no server hop.

WebRTC handles media capture (camera/microphone), encoding, NAT traversal (STUN/TURN), and adaptive bitrate. It can also carry arbitrary data via Data Channels.

Side-by-side comparison

• Topology: WebSocket = client→server; WebRTC = peer→peer (with signaling server for setup)
• Media: WebSocket carries text/binary only; WebRTC carries audio/video/data
• Latency: WebSocket = round-trip to server; WebRTC = direct, no server hop
• Scaling: WebSocket scales with server capacity; WebRTC mesh caps at ~10 peers
• NAT traversal: WebSocket uses server (no NAT issues); WebRTC needs STUN/TURN
• Use WebSocket for: chat, notifications, presence, dashboards
• Use WebRTC for: voice/video calls, screen sharing, large file transfer

How they work together

In a real video-call app, you use both: WebSockets connect each client to the signaling server (for join/leave, offer/answer/ICE exchange, presence). WebRTC is used for the actual audio/video media once the peers are connected. openbnet uses exactly this pattern: WebSocket signaling server + WebRTC media.

// WebSocket: signaling (server-mediated)
const ws = new WebSocket('wss://openbnet.com/ss/ws?user_id=me');
ws.send(JSON.stringify({ type: 'join-room', payload: { room_id: 'call-42' } }));

// WebRTC: media (peer-to-peer, server not in path)
const pc = new RTCPeerConnection(config);
pc.ontrack = (e) => attachToVideoElement(e.streams[0]);
localStream.getTracks().forEach(t => pc.addTrack(t, localStream));

// The offer/answer exchange goes over WebSocket (signaling),
// but the audio/video goes peer-to-peer (WebRTC).

openbnet uses both

openbnet is WebSocket-native across all its APIs: the Signaling Server, Chat, and Streaming all use WebSocket connections. Video calling uses WebSocket for signaling and WebRTC for peer-to-peer media. This is the standard architecture for real-time communication apps, managed for you by openbnet.

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