@amatiasq/peer-connection

WebRTC without the ceremony: one class that owns the RTCPeerConnection, the ICE gathering and the local/remote descriptions. You still have to carry the offer and the answer between the two peers yourself — that is what a signalling server is for.

npm i --save @amatiasq/peer-connection

Usage

Add your tracks before creating the offer or the answer; they are what the description advertises.

import { PeerConnection } from '@amatiasq/peer-connection';

const conn = new PeerConnection();
conn.onTrackReceived.subscribe(event => play(event.streams[0]));

const stream = await navigator.mediaDevices.getUserMedia({ video: true });
conn.addStream(stream);

// Caller
const offer = await conn.createOffer({ offerToReceiveVideo: true });
conn.acceptAnswer(await sendToPeerSomehow(offer));

// Callee
const answer = await conn.acceptOffer(offerFromPeer, {
  offerToReceiveVideo: true,
});
await sendToPeerSomehow(answer);

createOffer and acceptOffer resolve once ICE gathering has finished, so what they hand you is a complete description, ready to send in one message. Gathering waits for every ICE server it was given, so it is capped: a server that stops answering delays a call by three seconds instead of hanging it forever.

Your own ICE servers

The default is a pair of public STUN servers, enough to cross a home NAT and nothing more. Anything else — your own STUN, a TURN server for the networks STUN cannot cross — goes in the constructor, which takes a plain RTCConfiguration:

const conn = new PeerConnection({
  iceServers: [{ urls: 'turn:turn.example.com', username: 'me', credential: 'x' }],
});

Changing the streams mid-call

There is no way to renegotiate here: conn.reset() closes the connection and leaves the instance ready for a fresh handshake, which you start again from scratch.