Intermittently failed to play audio over webrtc/turn

[Yossi]

Hi, I'm trying to pass audio from my server to the client via WCS.
Publish rtsp opus audio
Play webrtc over turn

I sometime succeed to play the audio and sometimes fail.
WCS version 5.2.945

I've attached flashphoner log from my last case.
In this case I'm publishing rtsp stream, failed on the first time I'm trying to play it ("Failed by ICE timeout"), and succeeded on the second attempt.
Stream name: hssId_BOS-C5-2-6-cradle_VIRTUAL_02-60E5F12E_H264

I appreciate your help.
Yossi

 

[Max]

Good day.
Please collect a full report as described here and send using this form. We will check.

 

[Max]

Please try to switsh off RTP bundle support

rtp_bundle=false

and add local interface to ICE candidates

rtc_ice_add_local_interface=true

Also please make sure you are connecting to WCS to play a stream with Force relay enabled:

Flashphoner.createSession({
    urlServer: url,
    mediaOptions: {
          "iceServers": [{
                'url': 'turn:wcs:443?transport=tcp',
                'credential': 'coM77EMrV7Cwhyan',
                'username': 'flashphoner'
          }],
          "iceTransportPolicy": "relay"
    }
});

 

[Yossi]

No change. I've added these properties (and restart the service) and validated that we use "relay" in the client code.

 

[Max]

Please provide SSH access to the server using this form.

 

[Max]

We checked your server. Unfortunately, we did not found in logs the stream mentioned in report:

but found another stream.
Anyway, we cannot reproduce the problem in our test environment, so we need to reproduce it on your server. Please provide a stream published on server 24/7, and allow to connect to TURN port and websocket port from outside. We try to reproduce the issue and check how connection is establishing.