LiveKit Integration for Video Conferencing on Your Website

LiveKit — an open-source WebRTC server with SFU architecture that provides low latency and scaling to thousands of participants. You get full control over data and interface, with no participant limits or hidden fees. However, on real projects we encounter typical problems: incorrect ICE server configuration, NAT issues, and suboptimal SFU architecture choices, leading to latency spikes of 5–10 seconds. Our experience with over 50 WebRTC integrations shows that proper LiveKit configuration solves these issues at the core.

How to Reduce Video Call Latency with LiveKit

SFU (Selective Forwarding Unit) architecture ensures latency below 200 ms even on weak channels. Unlike Jitsi, LiveKit uses optimized codecs (VP9, H.264) and adaptive bitrate. With 50 participants, latency difference reaches 3–5 times in favor of LiveKit. Moreover, you pay no licensing fees: the server cost for 1000 users is minimal.

How to Integrate LiveKit: Case Study of an Educational Platform

We recently implemented LiveKit for a client — an educational platform with 5000 users. The original architecture used Jitsi, but after migrating to LiveKit, latency dropped from 800 ms to 150 ms, and server costs decreased by 40%. We used LiveKit Server 1.5 on Docker, Node.js backend with LiveKit SDK, and React 18 frontend with @livekit/components-react.

Installing LiveKit Server

# docker-compose.yml
services:
  livekit:
    image: livekit/livekit-server:latest
    command: --dev
    ports:
      - "7880:7880"
      - "7881:7881"
      - "7882:7882/udp"
    environment:
      LIVEKIT_KEYS: "APIkey: APIsecret"

For production, use a config with external IP and TURN servers:

# livekit.yaml
port: 7880
rtc:
  tcp_port: 7881
  udp_port: 7882
  use_external_ip: true
keys:
  APIkey: APIsecret

How to Integrate LiveKit into a React App?

Backend: token generation

import { AccessToken, RoomServiceClient } from 'livekit-server-sdk';

const roomService = new RoomServiceClient(
  process.env.LIVEKIT_URL,
  process.env.LIVEKIT_API_KEY,
  process.env.LIVEKIT_API_SECRET
);

await roomService.createRoom({
  name: `room-${meetingId}`,
  maxParticipants: 50,
  emptyTimeout: 10 * 60
});

function generateToken(roomName: string, participant: Participant): string {
  const at = new AccessToken(
    process.env.LIVEKIT_API_KEY,
    process.env.LIVEKIT_API_SECRET,
    { identity: participant.id, name: participant.name, ttl: '2h' }
  );
  at.addGrant({ roomJoin: true, room: roomName, canPublish: true, canSubscribe: true, canPublishData: true });
  return at.toJwt();
}

app.post('/api/meetings/:id/join', authenticate, async (req, res) => {
  const meeting = await meetingRepo.findById(req.params.id);
  if (!meeting) return res.status(404).json({ error: 'Meeting not found' });
  const token = generateToken(`room-${meeting.id}`, { id: req.user.id, name: req.user.displayName });
  res.json({ token, serverUrl: process.env.LIVEKIT_URL });
});

Client (React)

import { LiveKitRoom, VideoConference, RoomAudioRenderer } from '@livekit/components-react';

function VideoRoom({ roomId }) {
  const [token, setToken] = useState(null);
  const [serverUrl, setServerUrl] = useState('');

  useEffect(() => {
    fetch(`/api/meetings/${roomId}/join`, { method: 'POST' })
      .then(r => r.json())
      .then(({ token, serverUrl }) => { setToken(token); setServerUrl(serverUrl); });
  }, [roomId]);

  if (!token) return <div>Connecting...</div>;

  return (
    <LiveKitRoom token={token} serverUrl={serverUrl} connect video audio>
      <VideoConference />
      <RoomAudioRenderer />
    </LiveKitRoom>
  );
}

This is the minimal set. For a custom interface, use useTracks and GridLayout.

Recording Meetings

import { EgressClient, EncodedFileType } from 'livekit-server-sdk';

const egressClient = new EgressClient(
  process.env.LIVEKIT_URL,
  process.env.LIVEKIT_API_KEY,
  process.env.LIVEKIT_API_SECRET
);

const egress = await egressClient.startRoomCompositeEgress(
  `room-${meetingId}`,
  { file: { fileType: EncodedFileType.MP4, s3: { accessKey: process.env.AWS_ACCESS_KEY, secret: process.env.AWS_SECRET, bucket: 'recordings', key: `meetings/${meetingId}/${Date.now()}.mp4` } } }
);

Comparison of LiveKit with Alternatives

LiveKit is 3 times faster in latency than Jitsi and has no hidden costs.

How to Achieve Stable Operation with 1000 Participants?

For 1000+ participants, use SFU clustering. LiveKit supports horizontal scaling via Redis and NATS. Configure TURN servers (e.g., Coturn) for NAT traversal. Important: choose servers in multiple regions for low latency. Also enable adaptive bitrate and use the AdaptiveStream option on the client.

LiveKit Features: Version Comparison

Process of Work

What is Included in the Integration

• Setup of LiveKit Server on your or our server (Docker).
• Development of a backend module for token generation and room management.
• Integration of React components from @livekit/components-react with optional custom UI.
• Recording setup via Egress API (S3, MinIO).
• Documentation and 2 weeks of support after delivery.
• 30-day guarantee on stable video call operation.

Timeline

• Basic integration (server + tokens + React UI) — from 5 to 10 business days.
• Extended integration (custom UI, recording, admin panel) — from 15 to 25 business days.

LiveKit also supports end-to-end encryption and JWT authentication. In our integration, we configure security by default.

How to Order Integration?

Contact us for a free assessment of your project. We will analyze the requirements and offer the optimal solution. Get a consultation — write to Telegram or email. Implementing LiveKit pays off due to reduced infrastructure costs from the first month. Order LiveKit integration for your project today.

Development of Real-Time Systems: WebRTC, SSE, WebSocket

We know how painful it is when polling kills the server. One of our projects—an online auction platform—used polling every 2 seconds. Under a load of 400 participants, the server received 12,000 HTTP requests per minute for a single bid. 90% of responses were empty. After switching to WebSocket, the load dropped 15 times, saving approximately $3,000 per month on server costs. Order custom real‑time functions development—get a ready solution with a stability guarantee.

Implementing real‑time in production is not just a library. We design the architecture for load, scenarios, and budget. Below is a breakdown of key solutions with examples.

Choosing the Right Real-Time Transport for Your Project

Three Real-Time Transports: When to Choose Which

Server‑Sent Events work over regular HTTP/1.1 or HTTP/2. The browser opens a connection, the server keeps it open and pushes events in text/event-stream format. Automatic reconnection is built-in—no need for reconnect logic. Limitation: server → client only. Ideal for notifications, progress of long tasks, live feeds.

WebSocket is a full‑duplex channel after an HTTP Upgrade handshake. Browser and server exchange frames in both directions. Suitable for chats, collaborative editing, games, trading terminals. Requires separate reconnect logic and heartbeat (ping/pong every 30 seconds, otherwise NAT tables close the connection). The WebSocket protocol enables full‑duplex communication with minimal overhead (RFC 6455).

WebRTC is peer‑to‑peer audio/video and data directly between browsers, bypassing the server. A server is needed only for signaling (STUN/TURN for NAT traversal). A TURN server is required in 20–30% of cases (corporate networks, symmetric NAT). For a telemedicine service, we implemented WebRTC: audio latency dropped from 800 ms (via relay) to 50 ms—a 16‑fold improvement. The TURN server was needed only for 15% of sessions, saving significant traffic costs.

How to Properly Choose a Transport: Step-by-Step Guide

1. Determine the data exchange scenario: unidirectional (server → client) — SSE; bidirectional with low latency — WebSocket; audio/video — WebRTC.
2. Evaluate latency requirements. If below 500 ms is acceptable — SSE; for below 100 ms and bidirectional — WebSocket; for below 50 ms and P2P — WebRTC.
3. Check the infrastructure budget. SSE uses regular HTTP servers, WebSocket requires keeping connections in memory, WebRTC may require a TURN server (from a certain cost per TB of traffic).
4. Consider scaling: for 100k+ connections, consider a WebSocket gateway (Centrifugo, Pushpin).

What Is CRDT and How Is It Better Than Operational Transformation?

Collaborative editing is not just "whoever writes last wins". Without a conflict merging algorithm, two users insert text at position 45; the first saves—the position shifts; the second saves on top—the operation applies to an outdated state. Text gets duplicated or lost.

OT (Operational Transformation) requires a server to resolve conflicts; CRDT (Conflict‑free Replicated Data Types) works without a central coordinator. Yjs is the most mature CRDT library for the browser. It integrates with ProseMirror, TipTap, CodeMirror, Monaco Editor. CRDT (Yjs) is 5 times faster than OT for concurrent editing under high load.

Library comparison for collaborative editing

Issue: the Yjs document size grows due to operation history. Periodic garbage collection is needed—snapshot the document and clean old operations. Without it, a document worked on for a year may weigh 50 MB.

const ws = new WebSocket('wss://example.com');
let pingInterval;

ws.on('open', () => {
  pingInterval = setInterval(() => {
    ws.ping();
    setTimeout(() => {
      if (ws.readyState === WebSocket.OPEN) ws.terminate();
    }, 5000);
  }, 25000);
});

ws.on('close', () => clearInterval(pingInterval));

Common Mistakes in Real-Time Implementation and How to Avoid Them

Typical Mistakes in Real‑Time Implementation

Memory leak on the server—forgetting to remove the event handler when the connection closes. On Node.js, heap grows ~1 MB/hour. EventEmitter warns about 10+ listeners, but it's not always noticed.

Thundering herd on reconnect. The server goes down for 30 seconds, comes back—10,000 clients try to reconnect simultaneously. Exponential backoff with jitter is mandatory: delay = Math.min(baseDelay * 2^attempt + random(0, 1000), maxDelay).

Lack of connection lost indication. WebSocket doesn't always notify about disconnection (e.g., phone enters a tunnel). Heartbeat solves the problem.

Work Process

We start by choosing the transport for the scenarios—sometimes all three are needed in one project: SSE for system notifications, WebSocket for chat, WebRTC for video calls. We design the message protocol (JSON with type and payload, less often binary via MessagePack). We develop with race condition testing—this is not covered by unit tests.

Load testing with k6 + k6/experimental/websockets: we simulate 5,000 concurrent connections with a real pattern. Our engineers are certified in WebSocket and WebRTC, guaranteeing 99.9% stability.

What's Included in the Delivery

• Real‑time layer architecture (transport selection, message protocol)
• Implementation with load testing (k6, race condition scenarios)
• Backend integration via Redis Pub/Sub or similar bus
• Protocol and data schema documentation
• Team training
• Technical support for 2 weeks after launch

Why Centrifugo May Be More Cost-Effective Than Socket.io?

Socket.io is easier to set up (1–2 days), but Centrifugo built on Go handles 1M+ connections on a single node. For 100k concurrent clients, Centrifugo saves up to 40% on infrastructure costs, which translates to $2,000 per month compared to Socket.io. Get a consultation—we'll help you choose the stack for your load.

Timeline

• Basic WebSocket chat or notifications on top of existing API: 1–3 weeks.
• Collaborative editor with Yjs and persistence: 4–8 weeks.
• WebRTC video calls with recording: 6–12 weeks (significant part is integration with media server mediasoup or Janus).

Contact us to evaluate your project. Discuss your task with an engineer—we'll assess complexity and timeline individually.