Convert Webrtc track stream to URL (RTSP/UDP/RTP/Http) in Video tag - ffmpeg

Question

I am new in WebRTC and i have done client/server connection, from client i choose WebCam and post stream to server using Track and on Server side i am getting that track and assign track stream to video source. Everything till now fine but problem is now i include AI(Artificial Intelligence) and now i want to convert my track stream to URL maybe UDP/RTSP/RTP etc. So AI will use that URL for object detection. I don't know how we can convert track stream to URL.
Although there is a couple of packages like https://ffmpeg.org/ and RTP to Webrtc etc, i am using Nodejs, Socket.io and Webrtc, below you can check my client and server side code for getting and posting stream, i am following thi github code https://github.com/Basscord/webrtc-video-broadcast.
Now my main concern is to make track as a URL for video tag, is it possible or not or please suggest, any help would be appreciated.
Server.js
This is nodejs server code
const express = require("express");
const app = express();
let broadcaster;
const port = 4000;
const http = require("http");
const server = http.createServer(app);
const io = require("socket.io")(server);
app.use(express.static(__dirname + "/public"));
io.sockets.on("error", e => console.log(e));
io.sockets.on("connection", socket => {
socket.on("broadcaster", () => {
broadcaster = socket.id;
socket.broadcast.emit("broadcaster");
});
socket.on("watcher", () => {
socket.to(broadcaster).emit("watcher", socket.id);
});
socket.on("offer", (id, message) => {
socket.to(id).emit("offer", socket.id, message);
});
socket.on("answer", (id, message) => {
socket.to(id).emit("answer", socket.id, message);
});
socket.on("candidate", (id, message) => {
socket.to(id).emit("candidate", socket.id, message);
});
socket.on("disconnect", () => {
socket.to(broadcaster).emit("disconnectPeer", socket.id);
});
});
server.listen(port, () => console.log(`Server is running on port ${port}`));
Broadcast.js
This is the code for emit stream(track)
const peerConnections = {};
const config = {
iceServers: [
{
urls: ["stun:stun.l.google.com:19302"]
}
]
};
const socket = io.connect(window.location.origin);
socket.on("answer", (id, description) => {
peerConnections[id].setRemoteDescription(description);
});
socket.on("watcher", id => {
const peerConnection = new RTCPeerConnection(config);
peerConnections[id] = peerConnection;
let stream = videoElement.srcObject;
stream.getTracks().forEach(track => peerConnection.addTrack(track, stream));
peerConnection.onicecandidate = event => {
if (event.candidate) {
socket.emit("candidate", id, event.candidate);
}
};
peerConnection
.createOffer()
.then(sdp => peerConnection.setLocalDescription(sdp))
.then(() => {
socket.emit("offer", id, peerConnection.localDescription);
});
});
socket.on("candidate", (id, candidate) => {
peerConnections[id].addIceCandidate(new RTCIceCandidate(candidate));
});
socket.on("disconnectPeer", id => {
peerConnections[id].close();
delete peerConnections[id];
});
window.onunload = window.onbeforeunload = () => {
socket.close();
};
// Get camera and microphone
const videoElement = document.querySelector("video");
const audioSelect = document.querySelector("select#audioSource");
const videoSelect = document.querySelector("select#videoSource");
audioSelect.onchange = getStream;
videoSelect.onchange = getStream;
getStream()
.then(getDevices)
.then(gotDevices);
function getDevices() {
return navigator.mediaDevices.enumerateDevices();
}
function gotDevices(deviceInfos) {
window.deviceInfos = deviceInfos;
for (const deviceInfo of deviceInfos) {
const option = document.createElement("option");
option.value = deviceInfo.deviceId;
if (deviceInfo.kind === "audioinput") {
option.text = deviceInfo.label || `Microphone ${audioSelect.length + 1}`;
audioSelect.appendChild(option);
} else if (deviceInfo.kind === "videoinput") {
option.text = deviceInfo.label || `Camera ${videoSelect.length + 1}`;
videoSelect.appendChild(option);
}
}
}
function getStream() {
if (window.stream) {
window.stream.getTracks().forEach(track => {
track.stop();
});
}
const audioSource = audioSelect.value;
const videoSource = videoSelect.value;
const constraints = {
audio: { deviceId: audioSource ? { exact: audioSource } : undefined },
video: { deviceId: videoSource ? { exact: videoSource } : undefined }
};
return navigator.mediaDevices
.getUserMedia(constraints)
.then(gotStream)
.catch(handleError);
}
function gotStream(stream) {
window.stream = stream;
audioSelect.selectedIndex = [...audioSelect.options].findIndex(
option => option.text === stream.getAudioTracks()[0].label
);
videoSelect.selectedIndex = [...videoSelect.options].findIndex(
option => option.text === stream.getVideoTracks()[0].label
);
videoElement.srcObject = stream;
socket.emit("broadcaster");
}
function handleError(error) {
console.error("Error: ", error);
}
RemoteServer.js
This code is getting track and assign to video tag
let peerConnection;
const config = {
iceServers: [
{
urls: ["stun:stun.l.google.com:19302"]
}
]
};
const socket = io.connect(window.location.origin);
const video = document.querySelector("video");
socket.on("offer", (id, description) => {
peerConnection = new RTCPeerConnection(config);
peerConnection
.setRemoteDescription(description)
.then(() => peerConnection.createAnswer())
.then(sdp => peerConnection.setLocalDescription(sdp))
.then(() => {
socket.emit("answer", id, peerConnection.localDescription);
});
peerConnection.ontrack = event => {
video.srcObject = event.streams[0];
};
peerConnection.onicecandidate = event => {
if (event.candidate) {
socket.emit("candidate", id, event.candidate);
}
};
});
socket.on("candidate", (id, candidate) => {
peerConnection
.addIceCandidate(new RTCIceCandidate(candidate))
.catch(e => console.error(e));
});
socket.on("connect", () => {
socket.emit("watcher");
});
socket.on("broadcaster", () => {
socket.emit("watcher");
});
socket.on("disconnectPeer", () => {
peerConnection.close();
});
window.onunload = window.onbeforeunload = () => {
socket.close();
};

Answer 1

rtp-to-webrtc does exactly what you want.
Unfortunately you will need to run some sort of server to make this happen, it can’t all be in the browser. You could also upload via other protocols (captured via MediaRecorder) if you don’t want to use WebRTC.