Integrating Video Consultations into Your Website: SDKs, WebRTC, Jitsi

Doctors lose up to 30% of their time coordinating visits, and clients increasingly prefer online consultations. Integrating video consultations into your website solves this — for medical, legal, financial, and educational services. Over recent years, demand for embedded video consultations has grown by 40%, and many companies have already implemented such solutions. Mistakes in choosing a solution lead to delays and budget overruns.

In this article, we’ll cover the key technical aspects and share our integration experience. You’ll learn how to choose the right stack, avoid common pitfalls, and accelerate time to market. Conversion of a website with video consultations increases by 20–40%, and bounce rates decrease by 15–25%. Integrating video consultations is not just about conversion uplift — it also strengthens customer trust. We work with various stacks — from ready-made SDKs to custom WebRTC solutions — tailoring the best option for your project. Our engineers hold certifications from Daily.co and Twilio, and the average launch time is 2–4 weeks.

How to Choose the Right Video Consultation Solution

There are three paths: ready-made SDKs (Daily.co, Twilio, Agora), custom WebRTC stack, or open-source Jitsi. Each option differs in implementation time, data control, and cost. The comparison is shown in the table below:

If you need a quick launch in 2–4 weeks — choose an SDK. If security and full control are critical (medical, financial) — build WebRTC on mediasoup or Janus. Jitsi is a middle ground: free server, moderate control, integration in 2–3 weeks. At high loads, Jitsi can be 2–3 times cheaper than ready-made SDKs, but requires administration.

Ready-Made SDKs — Fast Deployment

Providers offer client SDKs and APIs for managing rooms. For example, Daily.co allows integrating video calls in a few hours. We use this approach for projects with tight deadlines.

// Daily.co integration
import Daily from '@daily-co/daily-js';

const callFrame = Daily.createFrame(document.getElementById('video-container'), {
    iframeStyle: { width: '100%', height: '600px' }
});

const room = await fetch('https://api.daily.co/v1/rooms', {
    method: 'POST',
    headers: { Authorization: `Bearer ${DAILY_API_KEY}` },
    body: JSON.stringify({
        name: `consultation-${consultationId}`,
        properties: {
            exp: Math.floor(Date.now() / 1000) + 3600,
            max_participants: 2
        }
    })
}).then(r => r.json());

await callFrame.join({ url: room.url, token: participantToken });

Custom WebRTC Stack — Full Control

When high confidentiality is required (medical, financial), we choose custom WebRTC. We deploy a server based on mediasoup or Janus, configure TURN/STUN, and implement signaling via WebSocket. This gives full control over quality and security, but requires more development time.

Jitsi Meet — Open-Source Solution

Jitsi Meet is a popular open-source option. It runs on your own server and supports recording, screen sharing, and chat. Example integration:

const jitsi = new JitsiMeetExternalAPI('meet.yourserver.ru', {
    roomName: `consultation-${consultationId}`,
    parentNode: document.getElementById('video-container'),
    userInfo: { displayName: user.name, email: user.email },
    configOverwrite: { startWithAudioMuted: false, startWithVideoMuted: false }
});

Integration Process: Step-by-Step Plan

1. Audit requirements and current architecture, determine peak loads.
2. Choose a solution: SDK, Jitsi, or custom WebRTC based on budget and needs.
3. Set up infrastructure: server, TURN/STUN for WebRTC, domains, and SSL.
4. Integrate video API and develop a consultation scheduler with notifications.
5. Test on various devices and connection types (3G, 4G, Wi-Fi).
6. Deploy to production and configure call quality monitoring.

Session Scheduling and Automatic Notifications

We design a database to store consultations: specialist ID, patient ID, scheduled time, status. 30 minutes before the start, the system sends a push notification or email with a unique entry link.

Why Integrate Video Consultations Now?

According to research by MarketsandMarkets, the telemedicine market is growing 25% annually. Websites with video chat increase conversion by 20–40%, and bounce rates drop by 15–25%. For clients, it’s convenience — consultation from anywhere. For business, it expands the audience and saves on rental costs. Our data shows that 70% of users prefer video consultations that can be recorded.

We guarantee stable operation even on weak internet (adaptive bitrate) and compliance with GDPR and Federal Law 152. The average payback period for integration is 3–4 months. Contact us for a preliminary assessment of your project — get a consultation tailored to your scenario.

Recording Consultations and Security

Most SDKs support video recording. Files can be stored on S3 or with the provider. Always obtain user consent for recording — display a notification in the interface before starting.

Access Management — Waiting Room

To prevent clients from entering the room before the specialist arrives, we implement a waiting mechanism. The user clicks "Enter" and sees a waiting message. The specialist receives a notification and starts the consultation. This is easily done via the knock mode in Daily.co or similar flags.

Adaptive Bitrate for Slow Internet

On poor connections, we automatically lower video quality. In WebRTC, you can force bitrate limits:

const sender = pc.getSenders().find(s => s.track?.kind === 'video');
const params = sender.getParameters();
params.encodings[0].maxBitrate = 500000;
await sender.setParameters(params);

Ready-made SDKs handle this automatically.

Common integration mistakes — not accounting for room lifetime and lack of reconnection mechanism on connection loss. To avoid this, set exp with a buffer and handle reconnection events. Another mistake is ignoring adaptive bitrate for users with poor internet.

What Our Work on Video Consultation Integration Includes

Within our cooperation, we provide:

• Audit of current architecture and loads.
• Selection of the optimal solution (SDK / WebRTC / Jitsi).
• Integration of video communication with your website or application.
• Setup of consultation scheduler and automatic notifications (email/push 30 minutes before).
• Testing on various devices and connection types.
• API documentation and user instructions.
• Training administrators to work with the system.
• Technical support after launch.

Timelines and Budget

Basic integration of Daily.co or Jitsi with scheduler — from 2 to 3 weeks. Full custom WebRTC stack — from 6 to 8 weeks. Cost is determined after analyzing your specific requirements. Contact us for an estimate tailored to your project.

WebRTC

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.