Add Video Calls to Your Site with Whereby Embedded
Adding video calls to your site is a critical requirement for telemedicine, EdTech, and SaaS. Manually building WebRTC with ICE candidates and STUN servers takes tens of hours, leads to N+1 requests, memory leaks, and browser incompatibility. Whereby Embedded solves this in one hour: a single <whereby-embed> gives you ready video calls with screen sharing, chat, and recording. Over 5 years we've integrated Whereby into 50+ projects—from doctor consultations to online schools. Our engineers are certified on the Whereby API and guarantee functionality across all devices. In one project, we reduced video delivery time from 3 weeks to 1 day, and infrastructure costs by 6x (saving over $30,000 per year). Below, we dive into the technical details that will save you time and budget.
Problems We Solve
WebRTC complexity. Self-built video requires setting up ICE servers, NAT handling, and codec management. One mistake and the call fails. Whereby handles it all: media servers, relay servers (TURN), and adaptive streaming. You get stable calls without headaches. Infrastructure savings—up to 90%.
Scaling. Growing load requires more server capacity for custom solutions. Whereby processes thousands of concurrent rooms automatically—you just keep your API key. This reduces infrastructure costs by 5–10x, and at 1,000 call hours per month, savings can reach $5,000.
Device support. The ready web component works correctly in all modern browsers (Chrome, Firefox, Safari, Edge) on desktop and mobile. Custom WebRTC needs individual testing for each browser and device.
Why Whereby Embedded Instead of Custom WebRTC?
| Feature |
Custom WebRTC |
Whereby Embedded |
| Development time |
2–4 weeks |
0.5–1 day |
| Server infrastructure |
Own TURN/STUN |
Ready (Whereby) |
| Browser testing |
Each browser |
Automatic |
| Scaling |
Manual |
Automatic |
| Maintenance cost |
High |
Fixed subscription from $99/month |
Note: As documented in the WebRTC API, implementation requires deep knowledge of network protocols. A ready component gives 10–20x time savings and eliminates stream handling headaches. Additionally, Whereby supports call recording and analytics—these require extra integrations in custom solutions.
Common Issues
Below are frequent mistakes during self-integration and how to avoid them.
| Mistake |
Symptom |
Solution |
| Missing CDN script |
Web component not rendering |
Add <script> before closing </body> |
| Incorrect endDate |
Rooms expire prematurely |
Set 1–2 hours buffer beyond scheduled end |
| Missing event handling |
UI not updating after call |
Subscribe to ready and leave |
| Ignoring hostRoomUrl |
Host cannot moderate |
Use hostRoomUrl for moderator entry |
Integration Process
The process takes 3 steps:
-
Create a room via API—we send a POST request to https://api.whereby.dev/v1/meetings with your API key. The room can be temporary (up to 2 hours) or persistent (with renewal). We automate room creation for your use case.
-
Embed the web component—add the script to <head> and place <whereby-embed> in the desired page section. For React/Next.js, declare the type and subscribe to ready and leave events. Setup takes minutes.
-
Customization—adjust display (minimal mode, disable premeeting screen, enable chat or screenshare). Optionally integrate with your CRM for call logging. All parameters are documented.
Example room creation via API
curl -X POST https://api.whereby.dev/v1/meetings \
-H "Authorization: Bearer YOUR_API_KEY" \
-H "Content-Type: application/json" \
-d '{"endDate": "2025-12-31T23:59:59Z", "roomMode": "normal"}'
Available Security Settings
Whereby Embedded provides flexible attributes to control functionality:
-
skip-media-permission-prompt—disable the camera permission request screen.
-
chat="off"—hide chat.
-
background="off"—disable background blur.
-
screenshare="off"—disable screen sharing.
For enterprise projects, audit logs and JWT authentication integration are available. This meets security and privacy requirements.
What's Included
- Whereby API integration—create and manage rooms.
- Web component embedding—correct operation in your framework.
- Documentation—all attributes and events explained.
- Access setup—API key and environment configuration.
- Training—how to manage rooms via admin panel.
- Support—2 weeks post-delivery (bug fixes).
Timeline
Basic integration (room creation + web component) takes 0.5–1 day. If custom logic is needed (automatic room creation on consultation booking, JWT authentication integration), up to 3 days. Timeline is fixed in contract.
Contact us for a free assessment of your project—we'll suggest the best solution and exact timeline. Order Whereby Embedded integration and get stable video calls as early as the next day.
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
- Determine the data exchange scenario: unidirectional (server → client) — SSE; bidirectional with low latency — WebSocket; audio/video — WebRTC.
- Evaluate latency requirements. If below 500 ms is acceptable — SSE; for below 100 ms and bidirectional — WebSocket; for below 50 ms and P2P — WebRTC.
- 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).
- Consider scaling: for 100k+ connections, consider a WebSocket gateway (Centrifugo, Pushpin).
| Transport |
Direction |
Latency |
Implementation Complexity |
Typical Scenarios |
| WebSocket |
Full duplex |
< 100 ms |
Medium |
Chats, games, trading |
| SSE |
Server → client only |
< 500 ms |
Low |
Notifications, progress feeds |
| WebRTC |
P2P audio/video/data |
< 50 ms |
High |
Video calls, file transfer |
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
| Library |
Algorithm |
Editor Support |
Complexity |
Performance |
| Yjs |
CRDT |
ProseMirror, TipTap, CodeMirror, Monaco |
Medium |
High (<10 ms at 100 ops) |
| ShareDB |
OT |
ProseMirror, Quill |
Medium |
Medium (requires merge server) |
| Automerge |
CRDT |
Any (RichText) |
High |
Good (but memory grows faster than Yjs) |
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.
WebSocket Heartbeat Example (Node.js)
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.
