How to Build a Video Calling App with React Native and WebRTC

Sophie F A — Wed, 15 Jul 2026 07:22:22 +0000

The WebRTC pipeline is where the interesting engineering lives. The UI around it — call setup screen, contact list, in-call controls, history log — is boilerplate you've built a hundred times. That's exactly the layer that RapidNative handles well: describe the screens you need, get production-ready React Native + Expo code you can drop straight into your Metro bundle.

For a video calling app, that's the pre-call lobby ("Start Call" / "Join Call" / recent contacts), the in-call overlay (avatar, timer, mute/video/end buttons, participant list for group calls), and the post-call summary. You can start from a PRD or sketch the flow and get a working Expo project in minutes, then wire in the react-native-webrtc layer described above.

The pattern generalizes: use AI to move fast on UI-heavy layers where you're implementing patterns everyone recognizes, and spend your engineering time on the parts that are genuinely hard — like the ICE candidate gathering that just failed on your test call. If you're building anything real-time, our writeups on how RapidNative delivers real-time React Native previews and how we built real-time collaboration for team app building cover the WebSocket patterns that pair nicely with WebRTC signaling.

Frequently Asked Questions

Does react-native-webrtc work with Expo Go?

No. react-native-webrtc ships native iOS and Android modules that Expo Go cannot load. You need to move to Expo Dev Client or the bare workflow and use EAS Build (or a local expo run:ios/expo run:android) to include the native code. The @config-plugins/react-native-webrtc plugin handles the required permissions automatically.

How much does a WebRTC video calling app cost to run?

For 1-to-1 calls, near zero — media flows peer-to-peer. Your only costs are the signaling server (a $5/month VPS handles thousands of concurrent calls) and TURN relay for the roughly 15–20% of calls that fall back to relayed mode. Budget $0.40 per GB of TURN traffic. Group calls with an SFU move more media through your servers and cost proportionally more.

Can I use WebRTC for one-to-many streaming?

WebRTC is optimized for two-way conversation. For one-to-many broadcast (like a webinar or Twitch-style stream), you can use an SFU, but at scale you'll pay more than a protocol like HLS or LL-HLS designed for that pattern. WebRTC shines when latency matters more than viewer count.

What's the difference between STUN and TURN?

STUN helps a peer discover its own public IP address so two peers behind NAT can attempt a direct connection. TURN is a fallback: when direct P2P fails (usually because both peers are behind symmetric NAT), TURN acts as a media relay. STUN is cheap and free servers exist; TURN uses your bandwidth and typically costs money.

Is WebRTC secure?

Yes. All WebRTC media is encrypted with DTLS-SRTP by default — it's not optional in the spec. You cannot send unencrypted media. The signaling channel is your responsibility; use WSS (secure WebSockets) or HTTPS for whatever you build.

Wrapping Up

You now have every piece needed for a production-quality react native video calling app: a signaling server, a peer connection lifecycle, media capture, offer/answer exchange, ICE handling, and the standard call controls. Add a TURN server, wire the UI to your app's state management, and you're 80% of the way to shipping.

The hardest part of WebRTC isn't the API — it's the network. Test on real cellular connections, log ICE state transitions, and instrument connection failures early. Everything else is UI polish, and that's the part AI is happiest to write for you.

Ready to build? Start with RapidNative to spin up the app shell around your WebRTC layer in minutes, or browse more React Native tutorials on the blog.

DEV Community: Sophie F A