{"id":1010,"date":"2026-05-07T09:44:52","date_gmt":"2026-05-07T02:44:52","guid":{"rendered":"https:\/\/liveapi.com\/blog\/react-native-webrtc\/"},"modified":"2026-05-07T10:20:26","modified_gmt":"2026-05-07T03:20:26","slug":"react-native-webrtc","status":"publish","type":"post","link":"https:\/\/liveapi.com\/blog\/react-native-webrtc\/","title":{"rendered":"React Native WebRTC: How It Works, Setup, and Code Examples"},"content":{"rendered":"Reading Time: <\/span> 10<\/span> minutes<\/span><\/span>

If you’re building a mobile app that needs real-time video, voice, or data, you’ve almost certainly run into react-native-webrtc<\/a>. It’s the de facto bridge between Google’s WebRTC stack and React Native \u2014 the same underlying engine that powers video calls in Chrome, Firefox, and Safari, wrapped in a JavaScript API you can call from your mobile app.<\/p>\n

But “install a library and call getUserMedia<\/code>” hides a lot of complexity. You still need to handle iOS and Android permissions, set up a signaling server, configure STUN and TURN, manage peer connections across screens, and figure out what happens when your app needs to scale past two users on a Wi-Fi network.<\/p>\n

This guide walks through the full picture: what React Native WebRTC is, how the architecture works, how to install and configure it on both platforms, a basic video-call code walkthrough, and the production patterns most demos skip. By the end you’ll know exactly when to ship a peer-to-peer mobile call with react-native-webrtc<\/code> and when to put a server in the middle.<\/p>\n

What Is React Native WebRTC?<\/h2>\n

React Native WebRTC is a native module that exposes the WebRTC<\/a> protocol stack to React Native apps through a JavaScript API that mirrors the browser WebRTC spec. The package is published on npm as react-native-webrtc<\/code> and is maintained by the open-source react-native-webrtc organization on GitHub.<\/p>\n

The library wraps Google’s libwebrtc \u2014 the same C++ implementation used by Chromium \u2014 and surfaces it on iOS, Android, and tvOS. Your React Native code calls familiar APIs like mediaDevices.getUserMedia()<\/code>, RTCPeerConnection<\/code>, and RTCIceCandidate<\/code>, and the module forwards those calls down to native iOS or Android WebRTC bindings. As of recent releases the package ships with WebRTC M124 and supports unified-plan SDP, simulcast, and software encode\/decode by default.<\/p>\n

Here’s how it compares to other ways of adding real-time video to a mobile app:<\/p>\n\n\n\n\n\n\n\n\n
Approach<\/th>\nLatency<\/th>\nCross-platform<\/th>\nCustom UI<\/th>\nServer cost<\/th>\nBest for<\/th>\n<\/tr>\n<\/thead>\n
react-native-webrtc<\/td>\n<500ms<\/td>\niOS + Android + tvOS<\/td>\nFull control<\/td>\nSignaling + STUN\/TURN<\/td>\nOne-to-one and small-group calls<\/td>\n<\/tr>\n
Native iOS\/Android SDKs<\/td>\n<500ms<\/td>\nPer-platform builds<\/td>\nFull control<\/td>\nSame as above<\/td>\nTeams with deep mobile expertise<\/td>\n<\/tr>\n
Commercial video SDK<\/td>\n<500ms<\/td>\niOS + Android + Web<\/td>\nLimited<\/td>\nBundled per-minute<\/td>\nFaster shipping, less control<\/td>\n<\/tr>\n
RTMP\/HLS streaming<\/td>\n2\u201330s<\/td>\nAnywhere a player runs<\/td>\nFull control<\/td>\nEncoding + CDN<\/td>\nOne-to-many broadcasts<\/td>\n<\/tr>\n<\/tbody>\n<\/table>\n

The trade-off is straightforward: react-native-webrtc<\/code> gives you sub-500ms latency and full control over the call experience, but you own the signaling, the TURN servers, and the scaling story. For a deeper comparison of the underlying protocols, see WebRTC vs HLS<\/a> and WebRTC vs RTMP<\/a>.<\/p>\n

How Does React Native WebRTC Work?<\/h2>\n

Under the hood, a React Native WebRTC call is the same dance every browser-based WebRTC app performs \u2014 your mobile device just plays the role of the browser. There are three moving parts: media capture, peer-to-peer negotiation, and a signaling channel that brokers the handshake.<\/p>\n

1. Media capture<\/h3>\n

Your app calls mediaDevices.getUserMedia({ audio, video })<\/code>. The native module asks iOS or Android for camera and microphone access, opens the hardware, and returns a MediaStream<\/code> object holding one or more MediaStreamTrack<\/code> instances. You attach those tracks to an on-screen RTCView<\/code> to show local preview.<\/p>\n

2. Peer connection setup<\/h3>\n

Each side of the call creates an RTCPeerConnection<\/code> and adds the local tracks. The peer connection is the object responsible for everything network-related \u2014 encoding, packetization, congestion control, encryption (DTLS-SRTP), and NAT traversal.<\/p>\n

To find a path through firewalls and home routers, the peer connection uses ICE (Interactive Connectivity Establishment). It contacts a STUN server to learn its public IP, gathers candidate addresses, and falls back to a TURN server to relay traffic when direct connectivity fails. If you want to go deeper here, STUN and TURN<\/a> are non-negotiable for any real-world deployment \u2014 most calls outside a single Wi-Fi network need them.<\/p>\n

3. Signaling<\/h3>\n

WebRTC deliberately doesn’t define how peers find each other. You build a signaling server \u2014 usually a small WebSocket or Socket.IO service \u2014 that ferries SDP offers, SDP answers, and ICE candidates between the two clients. Once the offer\/answer\/ICE exchange completes, the peer connection’s ontrack<\/code> event fires on both sides and media starts flowing directly between devices.<\/p>\n

The whole flow looks like:<\/p>\n

Caller                Signaling Server                Callee\r\n  |                          |                           |\r\n  |-- getUserMedia()         |                           |\r\n  |-- new RTCPeerConnection  |                           |\r\n  |-- createOffer ---------->|------------------------->|\r\n  |                          |       setRemoteDesc       |\r\n  |                          |       createAnswer        |\r\n  |<------------------------ |<-------- answer ---------|\r\n  |-- onicecandidate ------->|------ ICE candidates --->|\r\n  |<--------------------- ICE candidates -------------- |\r\n  |======== media (SRTP, peer-to-peer or via TURN) =====|\r\n<\/code><\/pre>\n
This is the same pattern a WebRTC server<\/a> coordinates at scale. For one-to-one mobile calls you can keep the signaling layer tiny \u2014 a Node.js process holding a WebSocket connection per client is enough.<\/p>\n
Key Features and Platform Support<\/h2>\n
The react-native-webrtc<\/code> package isn’t a thin wrapper. It exposes most of the modern browser WebRTC API surface and adds a few mobile-specific helpers.<\/p>\n
Supported features:<\/strong><\/p>\n