Hi everyone. First of all, thank you for the outstanding community. I’m implementing my first MediaSoup app to allow users to speak with each others. So far I have implemented everything both client and serverSide and I believe the flow is clear to me: I get the RtpCapabilities, I load the device, create the producer transport and connect, then I creeate che consumer transport and connect.
Whatching the reports and about:webrtc in firefox and my logs in the console it appears that the producers is flowing traffic (sending constantly 4KBps) but my inbound traffic for the other UDP connection is stuck at 0, so I guess this is why I have no audio in.
Before posting any code I just wanted to know what I might look at. I’m of course already looking at the consumer transport and connect methods, but I am clueless at how to debug audio tracks and TCP connections
To further expand my problem as I-m learning about UDP/TCP connections:
My Consumer ICE is succeded, nominated and selected, but the incoming bytes are stuck and the last package timestamp is still, therefore it appears to be up but without packets inbound. I guess I should expect it to disconnect since I have no traffic? But it’s up and still, so I’m really confused.
As imagined it was just a distraction mistake. After consume I wasn’t calling the consumerConnect and even there I was starting with the consumer paused >.< . Stupid me.