We tried jitsi for a while and found it did not perform well for screen sharing in large groups (which we use extensively). We switched to whereby.com (former appear.in) which is not free for group meetings (used to be though) but is much more reliable. Creating new meeting rooms works the same way as in jitsi
Hey there, any chance you remember when you tested this? We have recently fixed a number of problems with screen-sharing indeed!
Jitsi Meet: open-source video conferencing
91–96 of 96 posts
Re: Jitsi Meet: open-source video conferencing
#92Re: Jitsi Meet: open-source video conferencing
#93But I did want to toss this out there as an alternative to jit.si as it is functional to h.264 and sip clients and with some other software fronting it (like Kamailio) could be made to do SRTP and WebRTC with a html5/js sip client.
Re: Jitsi Meet: open-source video conferencing
#94Earlier quoted context omitted.
Are there any E2EE alternatives to it?
Nextcloud Talk is end-to-end encrypted, but not as easy to set up as Jitsi Meet. https://nextcloud.com/talk https://github.com/nextcloud/spreed
Re: Jitsi Meet: open-source video conferencing
#95Earlier quoted context omitted.
I think it's unwise to offer 1) proprietary solutions in the age of surveillance capitalism. 2) offer in-browser cryptography to ensure security. I don't see any end-to-end encryption elements in that. Your data gets essentially delivered to the service. Do we really want that? Haven't the companies like FB already shown you should never trust them with your data? I mean, I'm baffled to see people would go in circles…
Umm what phone are you using? I hope you realize that contrary to what you think Android isn't completely open source and some of the most key parts are owned by probably the biggest enemy of privacy on Earth.
Re: Jitsi Meet: open-source video conferencing
#96Earlier quoted context omitted.
Audio quality depends on jitter timing. Too low and you get dropouts, too high and you get uncanny valley with the delays and pauses. You should never have to twiddle these settings since internet routing is so dynamic. But things to look at include jitter sample size (20ms is a good start), codec selection (lower bandwidth), and server resource usage (more cpu or ram). I don’t use mumble but ran an asterix server fo…
In this case, it's mostly getting the levels set up and cutoff set up correctly since by default it cuts off when you aren't talking (and thankfully doesn't have AGC). There is a push to talk mode, but most people that I know don't use it.