Develop Reference

WebRtc works on Firefox but not on Chrome after changing signaling

I basicly Implemented a simple webrtc application where a client streams a video and then another client can connect and access the remote stream from each other.
The first implementation I did was with AgoraRTM signaling and it worked well in all browsers. After that I wanted to try to move to websockets as the signaling part.
When I finished moving to websockets, I noticed that it kept working at Firefox, but at Chrome and Edge (probably chromium browsers) the remoteStream doesn't show up (it shows up the first and second time, but stops working if I disconnect and connect a new client)
Here is my code:
main.js (frontend logic, webrtc)
let token = null;
let uid = String(Math.floor(Math.random() * 10000));
let queryString = window.location.search;
const urlSearch = new URLSearchParams(queryString);
const room = urlSearch.get("room");
if (!room) {
window.location = "lobby.html";
}
let client;
let channel;
let socket;
const constraints = {
video: {
width: { min: 640, ideal: 1920, max: 1920 },
height: { min: 480, ideal: 1080, max: 1920 },
aspectRatio: 1.777777778,
},
audio: false,
};
const servers = {
iceServers: [
{
urls: [
"stun:stun.l.google.com:19302",
"stun:stun1.l.google.com:19302",
"stun:stun2.l.google.com:19302",
"stun:stun3.l.google.com:19302",
"stun:stun4.l.google.com:19302",
],
},
],
};
const localVideoRef = document.getElementById("localVideo");
const remoteVideoRef = document.getElementById("remoteVideo");
let localStream;
let remoteStream;
let peerConnection;
const configureSignaling = async () => {
socket = await io.connect("http://localhost:4000");
socket.emit("join", { room, uid });
socket.on("MemberJoined", handleMemberJoined);
socket.on("MessageFromPeer", handleMessageFromPeer);
socket.on("MemberLeft", handleMemberLeft)
};
const handleMemberLeft = async () => {
remoteVideoRef.style.display = "none";
};
const handleMessageFromPeer = (m, uid) => {
const message = JSON.parse(m.text);
if(message.type !== "candidate") {
console.log('handleMessageFromPeer: ', message, uid)
}
if (message.type === "offer") {
createAnswer(uid, message.offer);
}
if (message.type === "answer") {
addAnswer(message.answer);
}
if (message.type === "candidate") {
if (peerConnection && peerConnection.currentRemoteDescription) {
peerConnection.addIceCandidate(message.candidate);
}
}
};
const createLocalStream = async () => {
localStream = await navigator.mediaDevices.getUserMedia(constraints);
localVideoRef.srcObject = localStream;
localVideoRef.play()
remoteVideoRef.play()
};
const init = async () => {
await configureSignaling();
await createLocalStream();
};
const handleMemberJoined = async (uid) => {
createOffer(uid);
};
let createOffer = async (uid) => {
await createPeerConnection(uid);
let offer = await peerConnection.createOffer();
console.log({ offer })
await peerConnection.setLocalDescription(offer);
console.log('localStream: ', offer)
socket.emit(
"sendMessageToPeer",
{ text: JSON.stringify({ type: "offer", offer: offer }) },
uid
);
};
let createPeerConnection = async (uid) => {
peerConnection = new RTCPeerConnection(servers);
remoteStream = new MediaStream();
remoteVideoRef.srcObject = remoteStream;
remoteVideoRef.style.display = "block";
remoteVideoRef.classList.add("remoteFrame");
if (!localStream) {
await createLocalStream();
}
localStream.getTracks().forEach((track) => {
peerConnection.addTrack(track, localStream);
});
peerConnection.ontrack = (event) => {
event.streams[0].getTracks().forEach((track) => {
remoteStream.addTrack(track);
});
};
peerConnection.onicecandidate = (event) => {
if (event.candidate) {
socket.emit(
"sendMessageToPeer",
{
text: JSON.stringify({
type: "candidate",
candidate: event.candidate,
}),
},
uid
);
}
};
};
let createAnswer = async (uid, offer) => {
await createPeerConnection(uid);
await peerConnection.setRemoteDescription(offer);
console.log('remoteStream: ', offer)
const answer = await peerConnection.createAnswer();
await peerConnection.setLocalDescription(answer);
console.log('localStream: ', answer)
socket.emit(
"sendMessageToPeer",
{ text: JSON.stringify({ type: "answer", answer: answer }) },
uid
);
};
let addAnswer = async (answer) => {
if (!peerConnection.currentRemoteDescription) {
peerConnection.setRemoteDescription(answer);
}
console.log(peerConnection)
};
let onLogout = async () => {
peerConnection.close()
remoteVideoRef.classList.remove("remoteFrame");
await socket.emit('onLeaveRoom', room)
};
let onToggleCamera = async () => {
const videoTrack = localStream
.getTracks()
.find((track) => track.kind === "video");
if (videoTrack.enabled) {
videoTrack.enabled = false;
document.getElementById("camera-btn").style.backgroundColor =
"rgb(255, 80, 80)";
} else {
videoTrack.enabled = true;
document.getElementById("camera-btn").style.backgroundColor =
"rgb(179, 102, 249, .9)";
}
};
let onToggleMic = async () => {
const audioTrack = localStream
.getTracks()
.find((track) => track.kind === "audio");
if (audioTrack.enabled) {
audioTrack.enabled = false;
document.getElementById("mic-btn").style.backgroundColor =
"rgb(255, 80, 80)";
} else {
audioTrack.enabled = true;
document.getElementById("mic-btn").style.backgroundColor =
"rgb(179, 102, 249, .9)";
}
};
window.addEventListener("beforeunload", onLogout);
init();
index.js (server logic, websockets)
const express = require("express");
const app = express();
const PORT = 4000;
const http = require("http").Server(app);
const cors = require("cors");
const users = []
app.use(cors());
const socketIO = require("socket.io")(http, {
cors: {
origin: "http://127.0.0.1:5501",
},
});
//Add this before the app.get() block
socketIO.on("connection", (socket) => {
socket.on('join', async ({room, uid}) => {
users.push(uid)
await socket.join(room);
socket.broadcast.to(room).emit('MemberJoined', uid)
})
socket.on("onLeaveRoom", async (room) => {
socket.broadcast.to(room).emit('MemberLeft')
})
socket.on("disconnect", async (room) => {
socket.broadcast.to(room).emit('MemberLeft')
})
socket.on('sendMessageToPeer', (data, uid) => {
socket.broadcast.emit('MessageFromPeer', data, uid )
});
});
app.get("/api", (req, res) => {
res.json({
message: "Hello world",
});
});
http.listen(PORT, () => {
console.log(`Server listening on ${PORT}`);
});
I tried a couple of things like:
Checking if the offer and the answer was sended correctly, just 1 time and with the correct SDP
Checked if there was a problem disconnecting from a socket that leads to have more clients in a room
Tried also to add localVideoRef.play() to check if there is an issue with the autoplay for chrome, since there was a similar thread at stackoverflow (Ended up not working)

webRTC not working when call initiated by FireFox

I am developing a simple webRTC application, using my own server for signaling.
the javascript code is as follow (I have removed the signaling process and unnecessary logic):
const configuration = {
iceServers: [
{
urls: [
"stun:stun1.l.google.com:19302",
"stun:stun2.l.google.com:19302",
],
},
],
iceCandidatePoolSize: 10,
};
const callerCandidatesString = "callerCandidates";
const calleeCandidatesString = "calleeCandidates";
var received_offer = null;
var offer = null;
var answer = null;
var peerConnection = null;
let localStream = null;
let remoteStream = null;
var constraints = {
optional: [],
mandatory: {
OfferToReceiveAudio: true,
OfferToReceiveVideo: true
}
}
async function startMedia(e) {
const localStream = await navigator.mediaDevices.getUserMedia({video: true, audio: true});
document.getElementById("video1").srcObject = localStream;
remoteStream = new MediaStream();
document.getElementById("video2").srcObject = remoteStream;
if (I am the caller) {
create_the_offer();
}
if (I am the callee) {
get_the_offer();
}
}
async function create_the_offer() {
peerConnection = new RTCPeerConnection(configuration);
registerPeerConnectionListeners();
localStream.getTracks().forEach(track => {
peerConnection.addTrack(track, localStream);
});
var offer = await peerConnection.createOffer(constraints);
peerConnection.setLocalDescription(offer);
peerConnection.onicecandidate = function(candidate) {
if (candidate.candidate == null) {
//save the offer in the server
--> offer: JSON.stringify(peerConnection.localDescription)},
}
}
check_if_there_is_an_answer();
}
async function get_the_offer() {
// --> retrieve the offer from the server, then
create_answer(offer_from_server);
}
async function create_answer(received_offer) {
console.log("Create PeerConnection with configuration: ", configuration);
peerConnection = new RTCPeerConnection(configuration);
registerPeerConnectionListeners();
localStream.getTracks().forEach(track => {
peerConnection.addTrack(track, localStream);
});
console.log('received offer:' + received_offer)
my_offer = new RTCSessionDescription(JSON.parse(received_offer));
peerConnection.setRemoteDescription(my_offer);
// collectIceCandidates(peerConnection, calleeCandidatesString, callerCandidatesString);
peerConnection.addEventListener("track", event => {
console.log("Got remote track:", event.streams[0]);
event.streams[0].getTracks().forEach(track => {
console.log("Add a track to the remoteStream:", track);
remoteStream.addTrack(track);
});
});
const answer = await peerConnection.createAnswer(constraints);
console.log("Created answer:", answer);
await peerConnection.setLocalDescription(answer);
peerConnection.onicecandidate = function (e) {
if (e.candidate == null) {
// --> send the answer to the server
}
}
function check_if_there_is_an_answer() {
// retrieve answer from server. this function is executed several times until the answer is received.
// when there is an aswer:
start_remote_connection(answer);
}
async function start_remote_connection(passed_answer) {
my_answer = new RTCSessionDescription(JSON.parse(passed_answer));
peerConnection.setRemoteDescription(my_answer);
peerConnection.addEventListener("track", event => {
console.log("Got remote track:", event.streams[0]);
event.streams[0].getTracks().forEach(track => {
console.log("Add a track to the remoteStream:", track);
remoteStream.addTrack(track);
});
console.log("stream remoto: " + JSON.stringify(remoteStream.getVideoTracks()));
});
document.getElementById("video1").srcObject = localStream;
document.getElementById("video2").srcObject = remoteStream;
}
async function hangUp(e) {
const tracks = document.getElementById("video1").srcObject.getTracks();
tracks.forEach(track => {
track.stop();
});
remoteStream.getTracks().forEach(track => track.stop());
peerConnection.close();
document.getElementById("video1").srcObject = null;
document.getElementById("video2").srcObject = null;
}
// collect ICE Candidates function below
async function collectIceCandidates(peerConnection, localName, remoteName) {
const candidatesCollection = null;
peerConnection.addEventListener("icecandidate", event => {
if (event.candidate) {
const json = event.candidate.toJSON();
candidatesCollection.add(json);
}
});
}
// collect ICE Candidates function above
function registerPeerConnectionListeners() {
peerConnection.addEventListener("icegatheringstatechange", () => {
console.log(
`ICE gathering state changed: ${peerConnection.iceGatheringState}`);
});
peerConnection.addEventListener("connectionstatechange", () => {
console.log(`Connection state change: ${peerConnection.connectionState}`);
});
peerConnection.addEventListener("signalingstatechange", () => {
console.log(`Signaling state change: ${peerConnection.signalingState}`);
});
peerConnection.addEventListener("iceconnectionstatechange ", () => {
console.log(
`ICE connection state change: ${peerConnection.iceConnectionState}`);
});
}
window.onload = startMedia();
If the caller uses chrome and the callee uses FireFox (on localhost, same PC) the code works fine and both users can share their screen.
output with chrome
If the caller uses FireFox and the callee uses Chrome (still on localhost) the code still works fine, but the connection is not established and users cannot see the screen of the other person. I get no error in the console.
output with FireFox
In particular, with FF I am not getting "connection state change: connecting" and then "connection state change: connected".
My guess is that FF and chrome manage the async/await differently, and somehow with FF some values are not ready when actually needed, but cannot figure out why ...
with safari (MacOS) it does not work!!
I then tried it between the computer (with chrome) and an Android phone (chrome browser). It worked the first time I test it, and then never again :(
Does anyone of you has a clue ?

Video stream with webRTC doesn't work if I try to reconnect

Basicly I am working in building a webRTC application using socket.io as signaling, I had previously agoraRTM and the app worked fine.
The app is quite simple, basicly I have the client streaming video and when I open a new tab I want to show a small screen with the local stream and a bigger one with the remote stream.
The application works the first time I enter, so, I open a first tab (I see my local Stream), I open the second one and I see both (remote and local stream).
The problems start happening when I close one of the tabs and try to duplicate the other tab again (basicaly leaving the room and joining again)
I tried a couple of things, like disconnecting the socket, leaving the room. I am not sure why it is happening. (I am learning websockets, it might be something that I am missing)
Here is my code:
client (main.js)
let token = null;
let uid = String(Math.floor(Math.random() * 10000));
let queryString = window.location.search;
const urlSearch = new URLSearchParams(queryString);
const room = urlSearch.get("room");
if (!room) {
window.location = "lobby.html";
}
let client;
let channel;
let socket;
const constraints = {
video: {
width: { min: 640, ideal: 1920, max: 1920 },
height: { min: 480, ideal: 1080, max: 1920 },
aspectRatio: 1.777777778,
},
audio: false,
};
const servers = {
iceServers: [
{
urls: [
"stun:stun.l.google.com:19302",
"stun:stun1.l.google.com:19302",
"stun:stun2.l.google.com:19302",
"stun:stun3.l.google.com:19302",
"stun:stun4.l.google.com:19302",
],
},
],
};
const localVideoRef = document.getElementById("localVideo");
const remoteVideoRef = document.getElementById("remoteVideo");
let localStream;
let remoteStream;
let peerConnection;
const configureSignaling = async () => {
socket = await io.connect("http://localhost:4000");
socket.emit("join", { room, uid });
socket.on("MemberJoined", handleMemberJoined);
socket.on("MessageFromPeer", handleMessageFromPeer);
socket.on("MemberLeft", handleMemberLeft)
};
const handleMemberLeft = async () => {
remoteVideoRef.style.display = "none";
};
const handleMessageFromPeer = (m, uid) => {
const message = JSON.parse(m.text);
if(message.type !== "candidate") {
console.log('handleMessageFromPeer: ', message, uid)
}
if (message.type === "offer") {
createAnswer(uid, message.offer);
}
if (message.type === "answer") {
addAnswer(message.answer);
}
if (message.type === "candidate") {
if (peerConnection && peerConnection.currentRemoteDescription) {
peerConnection.addIceCandidate(message.candidate);
}
}
};
const createLocalStream = async () => {
localStream = await navigator.mediaDevices.getUserMedia(constraints);
localVideoRef.srcObject = localStream;
};
const init = async () => {
await configureSignaling();
await createLocalStream();
};
const handleMemberJoined = async (uid) => {
createOffer(uid);
};
let createOffer = async (uid) => {
await createPeerConnection(uid);
let offer = await peerConnection.createOffer();
await peerConnection.setLocalDescription(offer);
socket.emit(
"sendMessageToPeer",
{ text: JSON.stringify({ type: "offer", offer: offer }) },
uid
);
};
let createPeerConnection = async (uid) => {
peerConnection = new RTCPeerConnection(servers);
remoteStream = new MediaStream();
remoteVideoRef.srcObject = remoteStream;
remoteVideoRef.style.display = "block";
remoteVideoRef.classList.add("remoteFrame");
if (!localStream) {
await createLocalStream();
}
localStream.getTracks().forEach((track) => {
peerConnection.addTrack(track, localStream);
});
peerConnection.ontrack = (event) => {
event.streams[0].getTracks().forEach((track) => {
remoteStream.addTrack(track);
});
};
peerConnection.onicecandidate = (event) => {
if (event.candidate) {
socket.emit(
"sendMessageToPeer",
{
text: JSON.stringify({
type: "candidate",
candidate: event.candidate,
}),
},
uid
);
}
};
};
let createAnswer = async (uid, offer) => {
await createPeerConnection(uid);
await peerConnection.setRemoteDescription(offer);
const answer = await peerConnection.createAnswer();
await peerConnection.setLocalDescription(answer);
socket.emit(
"sendMessageToPeer",
{ text: JSON.stringify({ type: "answer", answer: answer }) },
uid
);
};
let addAnswer = async (answer) => {
console.log(!peerConnection.currentRemoteDescription, answer)
if (!peerConnection.currentRemoteDescription) {
peerConnection.setRemoteDescription(answer);
}
};
let onLogout = async () => {
remoteVideoRef.classList.remove("remoteFrame");
await socket.emit('onLeaveRoom', room)
};
let onToggleCamera = async () => {
const videoTrack = localStream
.getTracks()
.find((track) => track.kind === "video");
if (videoTrack.enabled) {
videoTrack.enabled = false;
document.getElementById("camera-btn").style.backgroundColor =
"rgb(255, 80, 80)";
} else {
videoTrack.enabled = true;
document.getElementById("camera-btn").style.backgroundColor =
"rgb(179, 102, 249, .9)";
}
};
let onToggleMic = async () => {
const audioTrack = localStream
.getTracks()
.find((track) => track.kind === "audio");
if (audioTrack.enabled) {
audioTrack.enabled = false;
document.getElementById("mic-btn").style.backgroundColor =
"rgb(255, 80, 80)";
} else {
audioTrack.enabled = true;
document.getElementById("mic-btn").style.backgroundColor =
"rgb(179, 102, 249, .9)";
}
};
window.addEventListener("beforeunload", onLogout);
init();
server (index.js)
const express = require("express");
const app = express();
const PORT = 4000;
const http = require("http").Server(app);
const cors = require("cors");
app.use(cors());
const socketIO = require("socket.io")(http, {
cors: {
origin: "http://127.0.0.1:5501",
},
});
//Add this before the app.get() block
socketIO.on("connection", (socket) => {
console.log(`⚡: ${socket.id} user just connected!`);
console.log('CONNECTED')
socket.on('join', async ({room, uid}) => {
console.log({ room })
await socket.join(room);
socket.broadcast.emit('MemberJoined', uid)
})
socket.on("onLeaveRoom", async (room) => {
await socket.leave(room)
socket.broadcast.emit('MemberLeft')
})
socket.on("disconnect", async (room) => {
await socket.leave('123')
socket.broadcast.emit('MemberLeft')
})
socket.on('sendMessageToPeer', (data, uid) => socket.broadcast.emit('MessageFromPeer', data, uid));
});
app.get("/api", (req, res) => {
res.json({
message: "Hello world",
});
});
http.listen(PORT, () => {
console.log(`Server listening on ${PORT}`);
});
I tried a couple of things, like disconnecting the socket.io when the user leaves, leaving the room . I feel like I am missing something websockets related in this case. Would like to know if that is the case :)

Webrtc video broadcasting Viewer only works sometimes

I´m using Gabriel Tanners video broadcasting Tutorial on localhost it works great! Over Internet connection most of the times the viewer simply doesn't show anything. I spent hours testing and changing the lifecycle. Nothing works.
Here´s the Broadcaster code:
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);
}
The viewer:
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();
};
I´m very thankful for your help

Since the two devices are not in the same network, you will need a TURN server for the establishment of the connection. A TURN server is used to relay as a relay if the peer-to-peer connection fails, which seems to be happening in your case.
Since there are no public TURN servers out there, you will probably need to create your own. For that, I can recommend the following options:
Coturn - Has a lot of possible configuration options but is thereby harder to set up.
Pion TURN - Easier to set up but doesn't feature as many possibilities
If you want to learn more about the internal of WebRTC and how they work together, checkout out webrtcforthecurious.

Remote video not showing up on one end - WebRTC Video chat app

I'm new to webrtc and react. I'm developing a peer to peer video chat app. On the calling side, both remote video and local video shows up. But on the callee side, only local video shows up. I've been trying to find out where I'm doing wrong but not able to figure it out. One thing I noticed when I console.log peerconnection variable inside handle ice candidate function is, the 'connectionState' is still 'connecting' on the callee side.('connected' on the caller side).
EDIT: I have modified code for readability and using async await. Now I am getting "Failed to execute 'addIceCandidate' on 'RTCPeerConnection': Error processing ICE candidate" error.
EDIT 2: Modified code according to answer
Here is my ORIGINAL code
//refs for my video tag
const localVideoRef = useRef("");
const remoteVideoRef = useRef("");
//video elements
<video ref={remoteVideoRef} playsInline autoPlay className="remoteVideo"></video>
<video ref={localVideoRef} playsInline autoPlay muted className="localVideo"></video>
//button to start call
<button onClick={handleCall}>Call</button>
const handleCall = async () => {
createPeerConnection();
navigator.mediaDevices
.getUserMedia({
audio: true,
video: true
})
.then(function(localStream) {
let videoObj = localVideoRef.current;
videoObj.srcObject = localStream;
localStream
.getTracks()
.forEach(track => myPeerConnection.addTrack(track, localStream));
})
.catch("getUserMedia() error: ");
};
let myPeerConnection = null;
function createPeerConnection() {
myPeerConnection = new RTCPeerConnection({
iceServers: [
{
urls: "stun:stun2.1.google.com:19302"
}
]
});
myPeerConnection.onnegotiationneeded = handleNegotiationNeededEvent;
myPeerConnection.onicecandidate = handleICECandidateEvent;
myPeerConnection.ontrack = handleTrackEvent;
myPeerConnection.onsignalingstatechange = handleSignalingStateChangeEvent;
}
var isNegotiating = false;
const handleSignalingStateChangeEvent = () => {
isNegotiating = myPeerConnection.signalingState != "stable";
};
function handleNegotiationNeededEvent() {
if (isNegotiating) {
return;
}
isNegotiating = true;
myPeerConnection
.createOffer()
.then(function(offer) {
return myPeerConnection.setLocalDescription(offer);
})
.then(function() {
socket.emit("video-offer", {
from: authContext.user.name,
to: connectedTo,
sdp: myPeerConnection.localDescription
});
});
}
//checking if socket is initialized
if (socket) {
socket.on("gotOffer", data => {
handleVideoOfferMsg(data);
});
socket.on("gotCandidate", data => {
handleNewICECandidateMsg(data);
});
socket.on("gotAnswer", data => {
console.log("inside got answer");
handleGotAnswer(data);
});
}
function handleVideoOfferMsg(msg) {
createPeerConnection();
var desc = new RTCSessionDescription(msg.sdp);
myPeerConnection
.setRemoteDescription(desc)
.then(function() {
return navigator.mediaDevices.getUserMedia({
audio: true,
video: true
});
})
.then(function(stream) {
let localStream = stream;
let videoObj = localVideoRef.current;
videoObj.srcObject = stream;
localStream
.getTracks()
.forEach(track => myPeerConnection.addTrack(track, localStream));
})
.then(function() {
return myPeerConnection.createAnswer();
})
.then(function(answer) {
return myPeerConnection.setLocalDescription(answer);
})
.then(function() {
socket.emit("video-answer", {
from: authContext.user.name,
to: connectedTo,
sdp: myPeerConnection.localDescription
});
})
.catch("error");
}
async function handleGotAnswer(msg) {
if (!myPeerConnection) return;
// if (isNegotiating) return;
//I don't know why it's not working (no remote video on the caller side too) when I add above line. So, I am checking signaling state in the below line
if (myPeerConnection.signalingState == "stable") return;
await myPeerConnection.setRemoteDescription(
new RTCSessionDescription(msg.sdp)
);
}
function handleICECandidateEvent(event) {
if (!myPeerConnection) return;
if (isNegotiating) return;
if (event.candidate) {
socket.emit("candidate", {
to: connectedTo,
from: authContext.user.name,
candidate: event.candidate
});
}
}
function handleNewICECandidateMsg(msg) {
if (myPeerConnection.signalingState == "stable") return;
var candidate = new RTCIceCandidate(msg.candidate);
myPeerConnection.addIceCandidate(candidate).catch("error");
}
function handleTrackEvent(event) {
let videoObj = remoteVideoRef.current;
videoObj.srcObject = event.streams[0];
}
Here is my NEW code:
let pc1 = new RTCPeerConnection({
iceServers: [
{
urls: "stun:stun2.1.google.com:19302"
}
]
});
let pc2 = new RTCPeerConnection({
iceServers: [
{
urls: "stun:stun2.1.google.com:19302"
}
]
});
const handleCall = async () => {
let stream = await navigator.mediaDevices.getUserMedia({
audio: true,
video: true
});
let videoObj = localVideoRef.current;
videoObj.srcObject = stream;
let localStream = stream;
stream
.getTracks()
.forEach(async track => await pc1.addTrack(track, localStream));
pc1.onnegotiationneeded = async function() {
let offer = await pc1.createOffer();
await pc1.setLocalDescription(offer);
socket.emit("video-offer", {
from: authContext.user.name,
to: connectedTo,
sdp: pc1.localDescription
});
pc1.onicecandidate = function(event) {
if (event.candidate) {
socket.emit("candidate", {
pc: "pc1",
to: connectedTo,
from: authContext.user.name,
candidate: event.candidate
});
}
};
};
pc1.ontrack = function(event) {
let videoObj = remoteVideoRef.current;
videoObj.srcObject = event.streams[0];
};
};
//listening to socket emits from server related to video chat
if (socket) {
socket.on("gotOffer", data => {
//step 1 of callee
handleVideoOfferMsg(data);
});
socket.on("gotCandidate", data => {
handleNewICECandidateMsg(data);
});
socket.on("gotAnswer", data => {
handleGotAnswer(data);
});
}
async function handleVideoOfferMsg(msg) {
var desc = new RTCSessionDescription(msg.sdp);
await pc2.setRemoteDescription(desc);
let stream = await navigator.mediaDevices.getUserMedia({
audio: true,
video: true
});
let videoObj = localVideoRef.current;
videoObj.srcObject = stream;
let localStream = stream;
stream
.getTracks()
.forEach(async track => await pc2.addTrack(track, localStream));
let answer = await pc2.createAnswer();
await pc2.setLocalDescription(answer);
socket.emit("video-answer", {
from: authContext.user.name,
to: connectedTo,
sdp: pc2.localDescription
});
pc2.ontrack = function(event) {
let videoObj = remoteVideoRef.current;
videoObj.srcObject = event.streams[0];
};
pc2.onicecandidate = function(event) {
if (event.candidate) {
socket.emit("candidate", {
pc: "pc2",
to: connectedTo,
from: authContext.user.name,
candidate: event.candidate
});
}
};
}
async function handleGotAnswer(msg) {
if (pc1.signalingState == "stable") {
console.log("negotiating");
return;
}
await pc1.setRemoteDescription(new RTCSessionDescription(msg.sdp));
//INSERTED THIS
if (candidatesArray.length) {
candidatesArray.forEach(async msg => {
var candidate = new RTCIceCandidate(msg.candidate);
await pc1.addIceCandidate(candidate);
});
}
}
let candidatesArray = [];
async function handleNewICECandidateMsg(msg) {
if (msg.pc == "pc1") {
var candidate = new RTCIceCandidate(msg.candidate);
await pc2.addIceCandidate(candidate);
}
if (msg.pc == "pc2") {
try {
if (pc1.connectionState != "stable" && !pc1.remoteDescription) {
candidatesArray.push(msg);
return;
}
var candidate = new RTCIceCandidate(msg.candidate);
await pc1.addIceCandidate(candidate);
} catch (error) {
//this is where error is triggered.
console.log("error adding ice candidate: " + error);
}
}
}
I have not put my server side code, because I find no issues in it.
From what I understand, the error is because remotedescription is not being set when addicecandidate is called. May be because I am skipping setting remote description when signalingState is stable. But if I remove that line of code, I am getting another error - "Failed to set remote answer sdp: Called in wrong state: kStable"
Where am I going wrong?

Immediately after calling pc.setLocalDescription(), the PeerConnection will start emitting onicecandidate events, thanks to Trickle ICE. However, this means that maybe the first candidates are generated too fast and they get sent to the remote peer even before sending SDP Offer/Answer!
Maybe that's what happens in your case, and the first candidates are arriving too early from the other side. For this reason it's a good idea to check the PeerConnection signaling state: if it is stable and the remote description has been already set, then you can call pc.addIceCandidate(). If not, you store the candidate in a queue.
Later, when the remote description finally arrives, after setting it you manually add all the candidates that are waiting in the queue.
Here you can see code with this idea. Candidates are first queued, and later when the PeerConnection signaling state becomes stable, queued items are added.