peer2peer

Simple WebRTC wrapper for peer to peer connection setup in browsers.

It creates portable offer and contract objects that you move through your own signaling channel.

• Try the demo

• Read the specification

• Check the documentation

Compatibility

• Runtimes: modern browsers with WebRTC support;
• Module format: ESM and CJS.
• Required globals / APIs: RTCPeerConnection, RTCDataChannel, MediaStream, EventTarget, CustomEvent, crypto.randomUUID; media sharing also needs navigator.mediaDevices and document.
• TypeScript: bundled types.

Installation

npm install @sovereignbase/peer2peer
# or
pnpm add @sovereignbase/peer2peer
# or
yarn add @sovereignbase/peer2peer
# or
bun add @sovereignbase/peer2peer
# or
deno add jsr:@sovereignbase/peer2peer
# or
vlt install jsr:@sovereignbase/peer2peer
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Usage

Create a connection

import { P2PConnection } from '@sovereignbase/peer2peer'

type Message = {
  type: 'chat'
  text: string
}

/**
 * Package uses 4 Google STUN Servers by default,
 * you can add additional turn or stun servers.
 *
 * For example `https://developers.cloudflare.com/realtime/turn/`.
 *
 * WebRTC is end-to-end encrypted by default.
 * As long as you stick to standard STUN or TURN servers and
 * do not use something like a SFU, all data is e2ee via DTLS
 */

// Peer A
const offer = await P2PConnection.makeOffer()
// send `offer` to peer B using your own transport

// Peer B
const copies = await P2PConnection.acceptOffer(offer)
// send `copies.offeror` back to peer A
// keep `copies.offeree` on peer B

const peerA = new P2PConnection<Message>(copies.offeror)
const peerB = new P2PConnection<Message>(copies.offeree)

peerA.addEventListener('message', (event) => {
  console.log('peer A received', event.detail)
})

peerB.addEventListener('message', (event) => {
  console.log('peer B received', event.detail)
})

await peerA.ready()
await peerB.ready()

peerA.sendMessage({
  type: 'chat',
  text: 'hello from peer A',
})
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Share media

connection.addEventListener('camera', (event) => {
  document.body.append(event.detail) // HTMLVideoElement
})

connection.addEventListener('screen', (event) => {
  document.body.append(event.detail) // HTMLVideoElement
})

// Stream media
await connection.shareMicrophone()
await connection.shareCamera()
await connection.shareScreen()

// Stop streaming media
connection.stopSharingMicrophone()
connection.stopSharingCamera()
connection.stopSharingScreen()
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Local preview elements are exposed as:

• P2PConnection.localCameraVideoElement
• P2PConnection.localScreenVideoElement

Runtime behaviour

• Package does not provide signaling transport. You move Offer and Contract objects through your own channel, for example WebSocket, HTTP, QR, copy-paste or any other out-of-band transport.
• P2PConnection.makeOffer() and P2PConnection.acceptOffer() always include 4 public Google STUN servers by default and append any additional ICE servers you pass in.
• ready() resolves only after the underlying RTCDataChannel reaches the "open" state. If the peer connection fails, closes, or the channel errors first, it rejects with a typed P2PConnectionError.
• sendMessage() uses MessagePack encoding via @msgpack/msgpack. The "message" event receives the decoded payload as event.detail.
• Media sharing is lazy and shared. shareMicrophone() and shareCamera() reuse one cached getUserMedia() stream, and shareScreen() reuses one cached getDisplayMedia() stream.
• Local preview elements are also shared through the static P2PConnection.localCameraVideoElement and P2PConnection.localScreenVideoElement properties.
• Track additions trigger automatic in-band renegotiation over the existing data channel. The offeree side behaves as the polite peer during glare handling.

Errors

Failures throw P2PConnectionError. The code is stable and the message describes the specific failure site.

Supported error codes:

• CHANNEL_ERROR
• CHANNEL_CLOSED
• CHANNEL_NOT_AVAILABLE
• CONNECTION_NOT_READY
• UNKNOWN_PEER_CONTRACT
• MISSING_LOCAL_DESCRIPTION

Tests

npm test
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• Unit tests cover typed errors, contract validation, ICE setup, local description failures and readiness guards.
• Integration tests cover connection setup, message exchange, renegotiation, media sharing, fallback media behavior and failure paths.
• Browser E2E tests use Playwright with separate browser contexts and a WebSocket signaling server to verify real WebRTC connection establishment and isolated simultaneous peer pairs.
• Current automated coverage is 100% for statements, branches, functions and lines on the published runtime bundle.

Benchmarks

npm run bench
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Last measured on Node v22.14.0 (win32 x64):

Benchmark	Average	Min	Max
websocket-signaled ready()	242.39 ms	229.80 ms	284.30 ms
one-way sendMessage() delivery	5.36 ms	3.21 ms	8.50 ms
message throughput	186.72 msg/s	-	-

License

Apache-2.0