> [1] https://github.com/Glimesh/broadcast-box

This is really interesting! Any latency benchmarks comparing it to gstreamer UDP, P2P primarily?

I don't! But when adding WebRTC support to OBS I would see 120ms https://github.com/obsproject/obs-studio/pull/7926

This is me (Ohio) going to a Digital Ocean Droplet (NY) and back.

Thanks for the reply and apologies I don’t see noti for replies.

That’s really interesting, I must test it out sometime, 120ms is still pretty good. A previous test I did before was gstreamer (on an embedded nvidia jetson) sending to a laptop running OBS, latency as low as 40ms, but the problem is the need to use OBS on the viewer side (or gstreamer directly), but would love to integrate it within the webUI I made.

I am adding WebRTC sources to OBS. Does that makes things easier?

Then you can push+pull into Broadcast Box and use OBS on both sides!

It would definitely be a useful feature for sure, however, the way I plan to use it (or hopefully it will work with broadcast box) is to install it on the SBC nvidia and use it to send the low latency stream to the p2p webUI client, instead of how I did it last time where gstreamer on the SBC sent the stream to OBS (note that OBS had the gstreamer plugin). In that test, I got the best results for latency. Any other protocol resulted in ~2sec, which was bad for the application I needed. I did try webrtc back then, but it was not as efficient as using gstreamer directly as sink/source pipelines, so if I can have in my tests: Broadcast box (on SBC) -> webUI that would a big step forward as I need to get rid off OBS, if I tried that soon will definitely update you!

>When an Invite has been accepted you will a log message like this.

In the 'Make your phone call' part of the readme, I think you might be missing the word 'see' between 'you will' and 'a log'.

Fixed! Thank you :)