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Comment by kragen | Hacker News Reader

you know, almost every time i try to talk with my family on jitsi, there's some kind of glitch. they can't see my screen, or i can't see theirs, or they can see it but only in super low resolution, or they have the camera turned on but i can't see it, or we all get kicked off, or something. can broadcast box allow us to use obs studio (or some other free, open-source software) to stream to each other, without relying on a proprietary server? i don't need 100k+ clients, i'd be satisfied with reliable connectivity between 2–4 clients! and i could run a server outside of nat

i'm not going to get 120ms latency though. i'm in argentina, they're mostly in the usa, and i have 200+ milliseconds of latency over the internet to anything in the usa

if broadcast box isn't what i'm looking for, is there something else? i already know about zoom, google, and teams, but those all make us vulnerable to proprietary servers

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Having discovered this 15 minutes ago I can't really speak in absolutes, but firing up OBS and using their test server, I think this would allow you to do exactly what you want. It's trivial to get OBS set up to stream to this, and while I haven't tried it, the docs say that it supports multiple streams to the same stream key. I guess it would just show each stream side by side in the client?

I think you could run your own server, run an instance of the front end and distribute the instructions to setup OBS to your family.

Lots of options! More and more WebRTC SFUs are adding WHIP support. Check out https://galene.org/ it lets you do OBS in and has things like chat. Might be better if you are streaming to family and want chat. I would try and find a VPS that is best geographically located.

Cloudflare if you want this quick. Even though it is proprietary you have no vendor lock in, just change your WHIP URL in OBS if things aren’t going right.

apparently 'an sfu' is 'is a media server component capable of receiving multiple media streams and then deciding which of these media streams should be sent to which participants [of which the] main use is in supporting group calls and live streaming/broadcast scenarios.' according to https://bloggeek.me/webrtcglossary/sfu/

i suspect webrtc implementations in browsers are the source of many of the problems but i don't know how to debug that

In HLS the client decides what stream to use. I presume in WebRTC it’s a negotiation.

Now I have to read the protocol. I was thinking the SFU was just a proxy, but it sounds more like it also handles some signaling.

i think in webrtc your code can decide what stream to use or to transmit in a turing-complete way; it doesn't define anything like that

Jitsi is actually one of the better options. My only beef is that it got bloated with features I don't need.

i suspect that the problems with jitsi are maybe problems with webrtc in general, because i've experienced very similar things on the proprietary services. my limited experience with obs studio, though, has been flawless, and it seems to be the main daily driver for lots of people who make their living streaming?

is your jitsi self-hosted? if so, where is it located?

i'm using jitsi.milliways.info, but i don't host it

looks like that's hosted in DE - might not be the best latency? would try self hosting it on a box in Florida or something to try and get the best connection to you both

latency is totally tolerable, so i'm pretty sure the audio and video are flowing directly between our machines and not via .de

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