JitstiMeet - Self Hosted - Not working in UAE


I would like to thank all Jitsi community. It has been great learning for me and I could finally host my own video conferencing solution over Jitsi Meet.

It was working so well, I could build a pretty user community in UAE. All of sudden it stopped working. Strangely meet.jit.si is working fine without any issues.

I believe that my application also should work. I am getting error
‘[modules/RTC/BridgeChannel.js] <l._send>: Bridge Channel send: no opened channel.’

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No one can help me? I tried hosting a meeting with meet.jit.si. It is working well. But my self-hosted Jitsi meet instance is not working with three way calls. P2P is working fine.

Search for UDP/10000

thanks @emrah I tried my own turn server. It is working well sometimes. the behavior is so strange.

If a new user joins the call, he cannot see anyone. but the host can see him. No audio/video. I think I need to stop using Jitsi. It is way beyond my abilities.

Hi, What chrome version are you using, and when was the last you upgraded Jitsi?

Hi Girish,

I am using Version 86.0.4240.111 (Official Build) (x86_64). Chrome on Mac.

Jitsi is a fresh installation. I have been installing again and again to troubleshoot this issue.

Yesterday I followed this thread Turnserver Config to avoid "701" Problem

Strange, Video conferencing is working fine with the Chrome browser. I tried to connect with WiFi, du-LTE personal hotspot. Airtel UAE roaming. All are working fine.
However with my app only P2P is working. No group call at all. No idea what did I do wrong.

Here is my /etc/jitsi/videobridge/sip-communicator.properties


Is your hosting behind a NAT server? If yes, the try these advanced configurations by adding local, public IPs. P2P doesn’t involve JVB which is why would not give any issue.

I don’t know how to check if my server is behind NAT. I hosted on DigitalOcean Debian 10 server

Check the ifconfig. If the IP that you see is same as that of your self hosted server, this means you are not behind a NAT. If not same, you will have to add it in the sip-communicator props.

@mudila.jagan if you’re hosted on DigitalOcean, you’re almost certainly not behind a NAT.

Have you done a search here for 10000/UDP as emrah suggested? Since you mentioned you’ve tried reinstalling several times, I would say go with the straightforward installation (does not involve specific configuration of a TURN server). Make sure that’s working fine and then you can make changes from there. You always want to try and narrow down issues.