100ms Integration for Video Conferences on Your Website

Integration of 100ms for Video Conferences on Your Website

Typical problem: you embedded a Zoom iframe, but participants complain about delays, interface customization is impossible, and scaling is limited. 100ms solves this via an SDK with full control. 100ms — WebRTC-platform with React SDK, ready-to-use UI components, and flexible pricing. The key feature is Rooms and Templates: you create a template in the dashboard (role settings, recording, quality), then create rooms based on it. This eliminates manual rights management and allows scaling video conferences to hundreds of participants without overloading the backend.

We use 100ms to embed video conferences with full control: custom roles, recording, UI customization. No iframe hacks or Zoom limitations. If a hundred users on your site are trying to join a conference simultaneously, 100ms handles it with Rooms and Templates. Our clients save on average $5000 on developing their own WebRTC solution by switching to 100ms.

What Problems Does 100ms Solve?

100ms efficiently manages roles: you create a template with roles like host, guest, viewer. Each participant receives a token with the corresponding role. Cloud recording of conferences is enabled in the template, webhooks notify about start and end. WebRTC ensures real-time communication with minimal latency — confirmed by load testing. Roomkit allows you to override the interface to match your brand — colors, element layout, custom buttons.

Why Is 100ms Better Than a Custom WebRTC Solution?

Custom WebRTC requires implementing STUN/TURN servers, signaling, room management, and tokens. 100ms provides everything out of the box: a ready backend for creating rooms, token generation, and monitoring. This reduces development time by weeks and lowers the risk of errors. According to our data, integrating 100ms is 3 times faster than developing a custom WebRTC solution. The cost per video minute in 100ms starts at $0.004, significantly cheaper than alternatives (Twilio ~$0.008, Agora ~$0.007). Thus, you save both time and money.

How Does the Integration of 100ms on Your Site Work?

We build a server-side API for creating rooms and generating tokens. The client SDK (React) connects to the room and manages audio/video streams. Below are the key steps.

Creating a Room and Getting a Token

npm install @100mslive/roomkit-react @100mslive/react-sdk

// On the server — Management Token for API
import jwt from 'jsonwebtoken';

function generateManagementToken(): string {
  return jwt.sign(
    {
      access_key: process.env.HMS_ACCESS_KEY,
      type: 'management',
      version: 2,
      iat: Math.floor(Date.now() / 1000),
      nbf: Math.floor(Date.now() / 1000),
    },
    process.env.HMS_APP_SECRET!,
    { algorithm: 'HS256', expiresIn: '24h' }
  );
}

// Create a room
async function createRoom(name: string): Promise<string> {
  const mgmtToken = generateManagementToken();

  const response = await fetch('https://api.100ms.live/v2/rooms', {
    method: 'POST',
    headers: {
      'Authorization': `Bearer ${mgmtToken}`,
      'Content-Type': 'application/json',
    },
    body: JSON.stringify({
      name,
      template_id: process.env.HMS_TEMPLATE_ID,  // from 100ms Dashboard
    }),
  });

  const room = await response.json();
  return room.id;
}

// Get an App Token for a participant
async function getAppToken(roomId: string, userId: string, role: string): Promise<string> {
  const mgmtToken = generateManagementToken();

  const response = await fetch('https://api.100ms.live/v2/auth-token', {
    method: 'POST',
    headers: {
      'Authorization': `Bearer ${mgmtToken}`,
      'Content-Type': 'application/json',
    },
    body: JSON.stringify({
      room_id: roomId,
      user_id: userId,
      role,  // role from Template: 'host', 'guest', 'viewer'
    }),
  });

  const { token } = await response.json();
  return token;
}

// API endpoint
app.post('/api/video/join', authenticate, async (req, res) => {
  const { sessionId } = req.body;

  let session = await db.videoSessions.findById(sessionId);
  if (!session.hmsRoomId) {
    const roomId = await createRoom(`session-${sessionId}`);
    session = await db.videoSessions.update(sessionId, { hmsRoomId: roomId });
  }

  const isHost = session.hostId === req.user.id;
  const token = await getAppToken(session.hmsRoomId, req.user.id, isHost ? 'host' : 'guest');

  res.json({ token });
});

React Component with Roomkit

import {
  HMSRoomProvider,
  useHMSActions,
  useHMSStore,
  selectIsConnectedToRoom,
  selectPeers,
  selectLocalPeer,
  HMSPeer,
} from '@100mslive/react-sdk';

function VideoConference({ authToken, userName }: { authToken: string; userName: string }) {
  return (
    <HMSRoomProvider>
      <VideoRoom authToken={authToken} userName={userName} />
    </HMSRoomProvider>
  );
}

function VideoRoom({ authToken, userName }) {
  const hmsActions = useHMSActions();
  const isConnected = useHMSStore(selectIsConnectedToRoom);
  const peers = useHMSStore(selectPeers);
  const localPeer = useHMSStore(selectLocalPeer);

  useEffect(() => {
    hmsActions.join({ authToken, userName, settings: { isAudioMuted: false, isVideoMuted: false } });

    return () => {
      hmsActions.leave();
    };
  }, [authToken]);

  if (!isConnected) {
    return <div className="flex items-center justify-center h-64">Connecting...</div>;
  }

  return (
    <div>
      <div className="grid grid-cols-2 gap-4 p-4">
        {peers.map(peer => (
          <PeerTile key={peer.id} peer={peer} />
        ))}
      </div>

      <div className="flex gap-3 justify-center p-4">
        <button onClick={() => hmsActions.setLocalAudioEnabled(!localPeer?.audioEnabled)}
          className="p-3 rounded-full bg-gray-800 text-white">
          {localPeer?.audioEnabled ? '🎤' : '🔇'}
        </button>
        <button onClick={() => hmsActions.setLocalVideoEnabled(!localPeer?.videoEnabled)}
          className="p-3 rounded-full bg-gray-800 text-white">
          {localPeer?.videoEnabled ? '📹' : '📷'}
        </button>
        <button onClick={() => hmsActions.leave()}
          className="p-3 rounded-full bg-red-600 text-white">
          Leave
        </button>
      </div>
    </div>
  );
}

function PeerTile({ peer }: { peer: HMSPeer }) {
  const videoRef = useRef<HTMLVideoElement>(null);
  const hmsActions = useHMSActions();

  const videoTrack = useHMSStore(selectCameraStreamByPeerID(peer.id));

  useEffect(() => {
    if (videoRef.current && videoTrack) {
      hmsActions.attachVideo(videoTrack.id, videoRef.current);
    }
    return () => {
      if (videoRef.current && videoTrack) {
        hmsActions.detachVideo(videoTrack.id, videoRef.current);
      }
    };
  }, [videoTrack]);

  return (
    <div className="relative bg-gray-900 rounded-xl overflow-hidden aspect-video">
      <video ref={videoRef} autoPlay muted={peer.isLocal} playsInline className="w-full h-full object-cover" />
      <span className="absolute bottom-2 left-2 text-white text-sm bg-black/50 px-2 rounded">
        {peer.name} {peer.isLocal && '(you)'}
      </span>
    </div>
  );
}

What Is Included in the Integration?

Backend API: server endpoints for creating rooms, generating tokens, and managing templates. React components: connecting Roomkit, customizing UI to match your brand. Template setup: creating templates in the 100ms dashboard with desired roles and recording options. Testing: stability check with simultaneous connections of 50+ participants. Documentation: API description and deployment instructions. Post-support: consultations after integration.

Comparison of 100ms with Other WebRTC Solutions

100ms wins in customization and ease of integration, and holds a mid-range price position. We help choose the optimal solution for your tasks.

Stages of 100ms Integration

Timeline

Integration of 100ms with room creation, tokens, and React component — 2–3 days. With roles, recording, and webhooks — 4–5 days. Contact us to discuss your project details.

Why Choose Us?

Our team has over 5 years of experience in web development and integration of WebRTC solutions. We guarantee stability of video conferences under load and provide technical support after implementation. Certified specialists keep code up-to-date and track SDK updates. Core stack: React, TypeScript, Node.js, 100ms SDK.

Get a consultation — we will evaluate your project and offer the optimal solution for video conference integration. Get a consultation on 100ms integration. Contact us to discuss the details.

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.