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Why is there zero documentation?


#1

I use an Elastix phone system which has Openfire built in. I came across the Jitsi Videobridge Plugin and thought that it would make a great addition. I installed it and now I have a Jitsi Videobridge Settings Page in my Openfire server admin pages. So far, so good. Now what? I have Googled everything I can think of trying to find the slightest hint of how to get this working and there is NOTHING! Not a help file or a README that I can find! Couldn’t someone connected with this project or someone in the community who has successfully got this working take the time to tell the rest of us how? Please someone, give us a clue!


#2

Using elastix also.

You need a proper domain name (not ip) defined in openfire for it all to function correctly.

If you have openfire and jitsivideobridge plugin installed, go to (use Chrome):

https://your.domain.name:7443/jitsi/apps/ofmeet

Hopefully you should see yourself from your webcam, send another the address of the meeting room to test further


#3

All openfire plugins have read me files that you can access from the plugins admin web page of openfire. Otherwise, read the blogs

Jitsi Videobridge Plugin for Openfire ver 1.3.0
Jitsi Videobridge plugin ver 1.2

Jitsi Videobridge plugin ver 1.1.1


#4

Hello,

I have created a very basic how to set up guide based on my own experience setting it up while collecting data from the forums. Hopefully it can help someone complete the setup without having to search around for different things.

How to Configure a VideoConference Server using Ignite Openfire XMPP with Jitsi Videobridge Plugin and Sonicwall Router …


#5

Thank you for doing this. You are a star


#6

You are welcome. Thank you as well and thanks to the community that makes this product possible. It’s a great work!


#7

Hello Miguel,

did you also manage to get the SIP working?


#8

Negative. I did not try. We have an Asterisk PBX and for Asterisk/SIP, and I know there is an add-on integration module but I have not tinkered with it because I have not had any need for it. The in-band voice quality on the conference itself is superb, so I have just been using that. Works great!


#9

Sip has only been tested with few voip providers. I dont use Asterisk or any pbx, so dont know if it works or not. It works with voipcheap.com and voxbone inum