WebRTC How It Works – From History to Practical Use in Next.js and React Native

WebRTC (Web Real-Time Communication) is a technology that for several years has enabled audio and video streaming directly in the browser. It requires no additional plugins, while providing low latency and end‑to‑end encryption. In this article we will look at how WebRTC originated, what lies beneath its transport layer, and how to use it in modern projects based on Next.js and React Native.

History of the WebRTC protocol

The beginnings of WebRTC date back to 2011, when Google announced the Google Talk project as an open peer‑to‑peer communication platform. In 2012 a W3C working group was formed, and in 2015 the standard was officially adopted. Since then browsers – Chrome, Firefox, Safari and Edge – have implemented the full set of APIs, allowing developers to build video applications without intermediary streaming servers.

WebRTC architecture – what’s under the hood?

The basic WebRTC model is based on three components: getUserMedia (acquiring camera and microphone), RTCPeerConnection (P2P connection) and RTCDataChannel (binary data transfer). First, getUserMedia invokes the browser interface, which after obtaining user consent returns a MediaStream object. This stream is then passed to RTCPeerConnection, which negotiates the connection using the ICE (Interactive Connectivity Establishment) protocol.

ICE uses STUN (Session Traversal Utilities for NAT) and TURN (Traversal Using Relays around NAT) servers to discover public IP addresses and, when necessary, to tunnel traffic through a relay server. This allows a connection to be established even in complex NAT environments.

Once the connection is established, media are encoded with VP8/VP9 (video) and Opus (audio) codecs and transmitted over RTP (Real-time Transport Protocol) secured by DTLS (Datagram Transport Layer Security). The whole flow is encrypted, making WebRTC one of the most secure communication methods in the browser.

WebRTC vs. WebSocket – differences and collaboration

Although WebRTC and WebSocket are often confused, they serve different roles. WebSocket provides a bidirectional, full‑duplex TCP channel, ideal for exchanging small text messages or JSON. WebRTC, on the other hand, is optimized for large, continuous multimedia streams and operates over UDP, reducing latency.

In practice applications often combine both solutions: WebSocket is used for signaling – exchanging SDP (Session Description Protocol) offers and ICE candidates – while after negotiation the media traffic moves to WebRTC. This allows session control (e.g., text chat, notifications) without burdening the video connection.

WebRTC in browsers – support and limitations

All modern browsers support the core WebRTC APIs, but there are minor differences. Safari, as the newest player, only fully supports RTCPeerConnection from version 12, and some older Chrome versions required an experimental flag. Additionally, support for the AV1 codec is still limited, which can affect video quality at low bandwidth.

It is important to remember that browser privacy policies require explicit user consent to access the camera and microphone. Therefore every application must handle request denial and provide a clear interface explaining the purpose of data collection.

Practical example: WebRTC in Next.js

Below is a simple checklist that will help you integrate WebRTC into a Next.js project:

• Install simple-peer – a lightweight library that wraps WebRTC.
• Create an API endpoint (e.g., /api/signal) handling WebSocket or Server‑Sent Events for signal exchange.
• In a React component call navigator.mediaDevices.getUserMedia and pass the stream to new Peer({ initiator: true, stream }).
• Handle the signal, stream and error events – forward signals through the API.

import Peer from 'simple-peer'

export default function VideoChat(){
  const [stream, setStream] = React.useState(null)
  const [peer, setPeer] = React.useState(null)

  React.useEffect(()=>{
    navigator.mediaDevices.getUserMedia({video:true,audio:true})
      .then(s=>{setStream(s); const p=new Peer({initiator:true,stream:s}); setPeer(p)})
  },[])

  // przykładowa obsługa sygnałów przez WebSocket
  React.useEffect(()=>{
    if(!peer) return
    const ws=new WebSocket('/api/signal')
    ws.onmessage=e=>peer.signal(JSON.parse(e.data))
    peer.on('signal',data=>ws.send(JSON.stringify(data)))
    peer.on('stream',remote=>{document.getElementById('remoteVideo').srcObject=remote})
  },[peer])

  return 
    
    
  
}

This code shows a minimal skeleton: acquiring media, creating a connection, and exchanging signals via WebSocket. In production you should add TURN support, fallbacks, and a UI for managing connections.

WebRTC in React Native – key differences

On mobile platforms we don’t have native access to the browser API, so we use the react-native-webrtc library. The library exposes the same classes (MediaStream, RTCPeerConnection), but requires additional Android/iOS permissions and native dependency configuration (e.g., libjingle_peerconnection_so.so).

Just like in Next.js, signaling is most often done through WebSocket or Firebase Realtime Database. Below is a short checklist:

• Add CAMERA and RECORD_AUDIO permissions in AndroidManifest.xml and Info.plist.
• Install react-native-webrtc and link the native modules.
• Use mediaDevices.getUserMedia with parameters {video:true, audio:true}.
• Create an RTCPeerConnection and exchange offers over the chosen signaling channel.

“WebRTC is not just a technology – it’s a communication design philosophy where security and low latency are defaults, not add‑ons.”

Common errors and trade‑offs when implementing WebRTC

The most frequent issue is misconfiguring STUN/TURN servers. Without a public TURN server the connection may fail in corporate networks or behind double NAT. Another mistake is lacking renegotiation handling after network condition changes, which leads to stream breaks.

Trade‑offs appear when choosing codecs: VP8 is widely supported but doesn’t deliver the same high quality at low bitrate as H.264. On the other hand, H.264 requires a license and isn’t always available in Safari.

You should also monitor memory usage – not releasing MediaStream after a call ends causes leaks and performance drops, especially on mobile devices.

Conclusion and next steps

How WebRTC works – from history to practical implementations – shows that this technology is ready for building robust real‑time video applications. Thanks to the open standard and support in major browsers, you can focus on business logic rather than low‑level networking. If you’re planning a project that requires video or audio communication, consider WebRTC as the foundation, and we at Coderia.it will help you implement, optimize, and maintain it in production.