What Is Real-Time Communication (RTC)?

Real-time communication (RTC) is technology that lets people exchange information — voice, video, text, or data — with minimal perceptible delay. Email is not real-time. A phone call is. Here is what RTC means, the technologies behind it, and how it powers modern apps.

Definition

Real-time communication (RTC) is any technology that enables the exchange of information with a delay low enough that it feels instantaneous to humans. The threshold is typically under 200 milliseconds for interactive voice/video and under 500ms for text chat. Anything slower breaks the conversational rhythm.

RTC covers four pillars: voice calling, video calling, text chat, and live streaming. The underlying technologies are WebRTC (for peer-to-peer audio/video), WebSocket (for client-server messaging), and WebTransport/HTTP3 (emerging).

A brief history

The concept is old — telephony is real-time communication. But internet-based RTC is newer. The key milestones: SIP (1999) brought VoIP, WebSocket (2011) gave browsers persistent connections, and WebRTC (2011, standardized 2021) gave browsers native peer-to-peer audio/video. Before WebRTC, real-time video in a browser required Flash plugins or proprietary clients.

Today, RTC underpins Zoom, Discord, WhatsApp calls, Twitch, Slack, Google Meet, and any app with embedded calling or chat. The market has consolidated around API platforms (Twilio, Agora, Vonage, Daily, openbnet) so developers can add RTC without building media servers.

The components of an RTC system

• Signaling server — coordinates the connection handshake between peers
• Media server (SFU/MCU) — optional, for group calls or when P2P mesh is insufficient
• STUN/TURN servers — handle NAT traversal so peers behind firewalls can connect
• Chat/messaging layer — WebSocket server with persistence and pub/sub
• Recording and archiving — optional, for server-side media capture
• AI moderation — optional, for filtering content in real-time

Why latency matters

Human perception of "real-time" depends on the modality. For voice, latency over 150ms starts to feel like walkie-talkie (people talk over each other). For video, over 400ms feels laggy. For chat, under 1 second feels instant. The architecture choice (mesh vs SFU, P2P vs server-routed) directly determines latency: fewer hops = lower latency.

openbnet targets <100ms end-to-end for chat and uses WebRTC mesh for video (direct P2P = lowest possible latency for small groups).

Common use cases

• In-app voice and video calling (telehealth, education, social apps)
• Live streaming (events, gaming, creator economy, webinars)
• Real-time chat (customer support, community, gaming, collaboration)
• Collaborative editing (Google Docs-style shared editing)
• Real-time notifications (delivery tracking, market data)
• IoT telemetry (sensor dashboards, smart-home control)

Choosing an RTC platform

The three questions to ask: (1) Do you need peer-to-peer media (WebRTC) or just messaging (WebSocket)? (2) How many concurrent participants per session? (3) Do you need persistence, recording, moderation, or PSTN? The answers determine whether you need a mesh, SFU, or hybrid architecture — and which platform fits.

openbnet covers all four RTC pillars with one WebSocket + WebRTC stack: chat, video calling, live streaming, and signaling, plus AI moderation. Start on the free tier — no credit card — and explore the developer hub for quickstarts.

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