Group Video Conferencing on Website: WebRTC, LiveKit, React

Note: when 50 people participate in a webinar, audio latency becomes noticeable (over 500 ms), and video may freeze. A typical P2P solution with 50 peers creates 1225 connections — a heavy load on both client and network. We faced this: a client lost 20% of their audience due to drops. To avoid such problems, we use the SFU server LiveKit, which reduces latency to 150 ms and scales to 1000+ participants. We integrate group video conferences and embed video chat on websites with 3–50+ participants, microphone/camera management, breakout rooms, chat, screen sharing, and recording. This is an embeddable system built on React and LiveKit, ready for production. Such architecture ensures low latency and high quality even with weak internet. Our solutions are already used in educational platforms, telemedicine, and corporate portals.

LiveKit is an open platform with SFU architecture, providing 2x lower latency compared to Janus under the same load.

How to scale a video conference using SFU?

The choice of infrastructure depends on the number of simultaneous participants. Compare two approaches:

For most projects, LiveKit is optimal. We use it in production and guarantee stability.

Why does Simulcast improve video quality?

Simulcast — publishing multiple video streams of different quality (high/medium/low). LiveKit does this automatically. The subscriber receives the layer matching their bandwidth and tile size. Without Simulcast, video jerks or pixelates on weak networks. Our experience shows: Simulcast is 2.5 times more efficient than standard encoding on weak networks.

// Explicitly set quality for a specific participant
participant.setTrackSubscriptionPermissions(true, [
  { trackSid: videoTrack.sid, quality: VideoQuality.MEDIUM }
]);

How to implement participant management via API?

The host can mute a microphone or remove a participant via the server API. Example endpoints:

// Mute a specific participant (at host request)
app.post('/api/conferences/:roomName/mute/:participantId', authenticate, async (req, res) => {
  const conference = await db.conferences.findByRoomName(req.params.roomName);
  if (conference.hostId !== req.user.id) return res.status(403).end();

  await svc.mutePublishedTrack(
    req.params.roomName,
    req.params.participantId,
    'microphone-track',
    true
  );

  // Notify via Data channel
  await svc.sendData(
    req.params.roomName,
    Buffer.from(JSON.stringify({ type: 'muted_by_host', targetId: req.params.participantId })),
    [req.params.participantId]
  );

  res.json({ ok: true });
});

// Remove a participant
app.delete('/api/conferences/:roomName/participants/:participantId', authenticate, async (req, res) => {
  await svc.removeParticipant(req.params.roomName, req.params.participantId);
  res.json({ ok: true });
});

Implementing breakout rooms

Breakout rooms — separate rooms for small groups. We create them programmatically and send invitations via Data channel:

async function createBreakoutRooms(mainRoomName: string, groups: string[][]) {
  const breakoutRooms = await Promise.all(
    groups.map((group, i) =>
      createConference(`${mainRoomName}-breakout-${i}`, { maxParticipants: group.length + 2 })
    )
  );

  for (let i = 0; i < groups.length; i++) {
    for (const participantId of groups[i]) {
      const token = generateParticipantToken(
        `conf-${mainRoomName}-breakout-${i}`,
        participantId,
        '',
        'participant'
      );
      await svc.sendData(
        `conf-${mainRoomName}`,
        Buffer.from(JSON.stringify({ type: 'breakout_invite', token, roomIndex: i })),
        [participantId]
      );
    }
  }
}

How to integrate LiveKit in 5 steps

1. Install server and client packages:

npm install livekit-server-sdk  # server
npm install @livekit/components-react livekit-client  # client

2. Configure the server: create API keys and URL (e.g., wss://your-domain.com).
3. Create a room via RoomServiceClient.createRoom().
4. Generate a token for the participant with required permissions.
5. Connect the client: use LiveKitRoom from @livekit/components-react.

Example of creating a conference:

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

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

async function createConference(conferenceId: string, options: {
  maxParticipants?: number;
  enableRecording?: boolean;
}): Promise<void> {
  await svc.createRoom({
    name: `conf-${conferenceId}`,
    maxParticipants: options.maxParticipants ?? 50,
    emptyTimeout: 300,
    metadata: JSON.stringify({ conferenceId, createdAt: new Date().toISOString() }),
  } as RoomOptions);
}

function generateParticipantToken(
  roomName: string,
  userId: string,
  displayName: string,
  role: 'host' | 'moderator' | 'participant' | 'viewer'
): string {
  const at = new AccessToken(
    process.env.LIVEKIT_API_KEY!,
    process.env.LIVEKIT_API_SECRET!,
    { identity: userId, name: displayName, ttl: 4 * 60 * 60 }
  );

  at.addGrant({
    roomJoin: true,
    room: roomName,
    canPublish: role !== 'viewer',
    canSubscribe: true,
    canPublishData: true,
    roomAdmin: role === 'host',
    hidden: false,
  });

  return at.toJwt();
}

What is included in the work

• Conference architecture: SFU selection, Simulcast setup, Data channel design.
• UI development in React with LiveKit components: participant grid, control panel, chat, raise hand button.
• Server side: room creation and management, authentication, roles (host/moderator/participant/viewer).
• Integration of recording and breakout rooms (optional).
• API and deployment documentation.
• Load testing up to 50 participants.

Timelines and Cost

Basic group conference with LiveKit and React — from 1 week. With breakout rooms, raise hand, recording, and management — 2–3 weeks. Cost is calculated individually based on the scope of work and necessary customizations. Bandwidth savings thanks to Simulcast reach 40% compared to standard encoding. Contact us — we'll find the best solution for your budget. Get a consultation on implementation.

Typical mistakes and their solutions

We have over 5 years of experience in WebRTC development and have completed more than 50 projects for conferences and live streams. Contact us — we'll help embed video communication on your site.

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.