Ice server not working on vitxi vitalpbx 4

VitXi WebRTC
1

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)

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This disrupts our client’s operations.

Thanks

2

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:(

3

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,

4

Hi,

all

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relay

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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.

5

Could you please send your iceServers details in a private message so we can test them on our development servers?

6

Hi @maynor

Sorry sir, I have sent a private message, pls check.

1 Like
7

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>
1 Like
8

Hi Sir,

Can you update testing turn server, if yes i will shutdown my turn server.

Thanks

1 Like
9

Hi,

We will test the turn server today,

Regards,

10

We have been able to reproduce the problem. We will release a new version of VitXi in the middle of next week.

1 Like
11

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,

1 Like