WebRTC does not connect after deployed to VPS (TURN server added

Question

The app: I'm developing app that allows peers to connect to each other, I use mediasoup and WebRTC (SFU).

The problem: Everything works locally but when deployed to VPS (I use Vultr) peers can not connect. On VPS firewall ports are open for the range declared in mediasoup (tcp and udp).

I went through several questions here on stackoverflow but unfortunatelly none of it helps:

• set up TURN server: WebRTC connection does not establish after successful signaling How do I configure WebRTC to connect to my TURN server?

I use metered.ca, and I pass configuration on the client side:

producerTransportRef.current = deviceRef.current.createSendTransport({
                ...params,
                iceServers: [
                    {
                        urls: 'stun:stun.relay.metered.ca:80',
                    },
                    {
                        urls: 'turn:a.relay.metered.ca:80',
                        username: 'XXX',
                        credential: 'YYY',
                    },
                    {
                        urls: 'turn:a.relay.metered.ca:80?transport=tcp',
                        username: 'XXX',
                        credential: 'YYY',
                    },
                    {
                        urls: 'turn:a.relay.metered.ca:443',
                        username: 'XXX',
                        credential: 'YYY',
                    },
                    {
                        urls: 'turn:a.relay.metered.ca:443?transport=tcp',
                        username: 'XXX',
                        credential: 'YYY',
                    },
                ],
            });

params come from the server from defining mediasoup transport and are passed to the frontend where are merged with TURN server configs:

let transport = await router.createWebRtcTransport(webRtcTransport_options);
callback({
                // https://mediasoup.org/documentation/v3/mediasoup-client/api/#TransportOptions
                params: {
                    id: transport.id,
                    iceParameters: transport.iceParameters,
                    iceCandidates: transport.iceCandidates,
                    dtlsParameters: transport.dtlsParameters,
                },
            });

When I print the iceCandidate I get from the server there's my server ip address

When I run webrtc-internals in chrome I get:

I followed the error code:

701 error code - If no host candidate can reach the server, errorCode will be set to the value 701 which is outside the STUN error code range.

I noticed that the host candidate is in my local network (not sure if that's the problem)

I'd highly appreciate even suggestions on what to check next

Answer 1

Error code 701 here is due to Metered not responding to your binding_request (error code 701 indicates a binding_request timeout, the purpose of the binding_request is to check your public IP, essentially to create a srflx candidate). This is because you are using the "turn:" scheme instead of "stun:".

For Chrome, when you perform the trickle ice candidate check here, you can use Wireshark to capture the sent packets. You will see the binding_request being sent out, but there won't be a response from Metered. Instead, Metered will only respond to your allocate_request (basically to create a relay candidate). So, in my opinion, this error code is not the main issue here. I think you need to provide additional information to understand why you cannot establish a WebRTC connection.