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
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,