Stream Web Audio with WebRTC without asking for microphone - webrtc

Question

I want to stream audio from a web page to a local server, using WebRTC. That server will process that audio and will output it immediately to the user. I need real time.
My code is actually working. However I am asking the user for the microphone with getUserMedia, and I don't need that microphone. This is quite annoying. What can I do in order to stream the audio without having to ask the user for the microphone?
Thank you.
Here is a minimal working example (it is highly inspired by https://github.com/aiortc/aiortc/blob/main/examples/server/client.js). Only the last part with comments is interesting :
let webSocket = new WebSocket('wss://0.0.0.0:8080/ws');
const config = { sdpSemantics: 'unified-plan' }
const pc = new RTCPeerConnection(config);
webSocket.onmessage = (message) => {
const data = JSON.parse(message.data);
switch(data.type) {
case "answer":
pc.setRemoteDescription(data.answer)
break;
default:
break;
}
};
function negotiate() {
return pc.createOffer()
.then(function(offer) {
return pc.setLocalDescription(offer);
})
.then(function() {
return new Promise(function(resolve) {
if (pc.iceGatheringState === 'complete') {
resolve();
} else {
function checkState() {
if (pc.iceGatheringState === 'complete') {
pc.removeEventListener('icegatheringstatechange', checkState);
resolve();
}
}
pc.addEventListener('icegatheringstatechange', checkState);
}
});
})
.then(function() {
const offer = pc.localDescription;
webSocket.send(
JSON.stringify({
type: "offer",
offer: {
sdp: offer.sdp,
type: offer.type
}
})
);
})
}
// Preparing the oscillator
const audioCtx = new (window.AudioContext || window.webkitAudioContext)();
const oscillator = audioCtx.createOscillator();
const serverDestination = audioCtx.createMediaStreamDestination();
oscillator.connect(serverDestination);
// Asking for useless microphone
navigator.mediaDevices.getUserMedia({audio: true})
.then(() => {
return negotiate();
});
// Actual streaming
const stream = new MediaStream();
serverDestination.stream.getTracks().forEach((track) => {
pc.addTrack(track, stream);
})
// User pushes button to start the oscillator
function play() {
oscillator.start();
};

Answer 1

Just get rid of this:
// Asking for useless microphone
navigator.mediaDevices.getUserMedia({audio: true})
.then(() => {
return negotiate();
});
As you say, it's useless and not necessary. If you don't call getUserMedia(), the user won't be prompted to share their microphone. You can make WebRTC connections without this.
I suspect the problem you're running into is that your audio context is paused. If you call audioCtx.resume() when a user clicks a button, you'll be up and running. This is due to autoplay policy.

Answer 2

If you don't need user media, don't ask for it with getUserMedia in your code.