Dear Vitalpbx Team,
Can you check why ice server not working on V4? I migrate V3 to V4. On V3 it working.
Log Error on Firebug (V4)
This disrupts our client’s operations.
Thanks
Hi Vitalpbx,
Can you confirm whether this is a bug or not? If so, I would immediately rollback to V3 until you resolve this issue:(
Hello,
The message indicates that the second server is either not working or there is no connection.
Please test that ICE server on this page: ICE Server Test.
Best regards,
Hi,
all
relay
I think the problem is not with my turn server, because I have used this turn server on vitalpbx3 for almost 1 year while on vitalpbx4 an error appears. If you look at using trickle-ice, it runs normally.
I think you should check vitalpbx4 webrtc ice server & look at firebug, try connecting to ice server using username & password.
Could you please send your iceServers details in a private message so we can test them on our development servers?
Hi @maynor
Sorry sir, I have sent a private message, pls check.
Hi @maynor
Example multiple iceServer turn & turns with transports
<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="UTF-8">
<meta name="viewport" content="width=device-width, initial-scale=1.0">
<title>WebRTC ICE Servers with Transports Example</title>
</head>
<body>
<h1>WebRTC ICE Servers with Transports Example</h1>
<video id="localVideo" autoplay playsinline></video>
<video id="remoteVideo" autoplay playsinline></video>
<script>
const localVideo = document.getElementById('localVideo');
const remoteVideo = document.getElementById('remoteVideo');
let localStream;
let peerConnection;
const iceServers = {
iceServers: [
// STUN servers
{
urls: 'stun:stun.l.google.com:19302'
},
{
urls: 'stun:stun1.l.google.com:19302'
},
// TURN servers with transport protocols
{
urls: [
'turn:your-turn-server.com:3478?transport=udp',
'turn:your-turn-server.com:3478?transport=tcp',
'turns:your-turn-server.com:5349?transport=tcp'
],
username: 'your-username',
credential: 'your-password'
}
]
};
async function start() {
localStream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true });
localVideo.srcObject = localStream;
peerConnection = new RTCPeerConnection(iceServers);
peerConnection.onicecandidate = (event) => {
if (event.candidate) {
console.log('New ICE candidate:', event.candidate);
}
};
peerConnection.ontrack = (event) => {
remoteVideo.srcObject = event.streams[0];
};
localStream.getTracks().forEach(track => {
peerConnection.addTrack(track, localStream);
});
const offer = await peerConnection.createOffer();
await peerConnection.setLocalDescription(offer);
// Simulate sending the offer to the remote peer
console.log('Local Offer:', offer);
// In a real application, you would send the offer to the remote peer via your signaling server
// and receive the answer from the remote peer in a similar way.
// Simulate receiving an answer from the remote peer
const remoteAnswer = new RTCSessionDescription({
type: 'answer',
sdp: 'your-answer-sdp' // This should be replaced with the actual SDP from the remote peer
});
await peerConnection.setRemoteDescription(remoteAnswer);
}
start();
</script>
</body>
</html>
Hi Sir,
Can you update testing turn server, if yes i will shutdown my turn server.
Thanks
Hi,
We will test the turn server today,
Regards,
We have been able to reproduce the problem. We will release a new version of VitXi in the middle of next week.
Hello,
A new version is now available, which addresses the issues reported in this post.
You can view all the details of the new version at the following link: VitXi WebRTC 4.2.0.0 Now Available
Best regards,