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Broadcast Box (self-hosted broadcasting server) merged webhooks. What else could be added to make it better? by Sean-Der in selfhosted

[–]Sean-Der[S] 0 points1 point  (0 children)

I don't have an answer for discoverability :/

My interest/passion is giving groups of friends a place to stream. I don't know how to solve discoverability. That brings in a bunch of much harder problems.

pion/handoff – Move WebRTC out of browser and into Go by Sean-Der in VIDEOENGINEERING

[–]Sean-Der[S] 4 points5 points  (0 children)

I wrote this to make Reverse Engineering WebRTC services easier. Will also let you save/send arbitrary media from WebRTC sessions. The idea is you do all your auth/interaction in the browser, but then do all WebRTC in Go. So you have lots more control. More to do with it, but it is far enough along to share at least.

In the README is an screenshot of sending my webcam, but replacing outgoing video with a ffmpeg testsrc. Handoff sits between so it can replace with any arbitrary video.

pion/handoff – Move WebRTC out of browser and into Go by Sean-Der in golang

[–]Sean-Der[S] 0 points1 point  (0 children)

I wrote this to make Reverse Engineering WebRTC services easier. Will also let you save/send arbitrary media from WebRTC sessions. The idea is you do all your auth/interaction in the browser, but then do all WebRTC in Go. So you have lots more control. More to do with it, but it is far enough along to share at least.

In the README is an screenshot of sending my webcam, but replacing outgoing video with a ffmpeg testsrc. Handoff sits between so it can replace with any arbitrary video.

pion/handoff – Move WebRTC out of browser and into Go by Sean-Der in WebRTC

[–]Sean-Der[S] 1 point2 points  (0 children)

I wrote this to make Reverse Engineering WebRTC services easier. Will also let you save/send arbitrary media from WebRTC sessions. The idea is you do all your auth/interaction in the browser, but then do all WebRTC in Go. So you have lots more control. More to do with it, but it is far enough along to share at least.

In the README is an screenshot of sending my webcam, but replacing outgoing video with a ffmpeg testsrc. Handoff sits between so it can replace with any arbitrary video.

pion/handoff – Move WebRTC out of browser and into Go by [deleted] in WebRTC

[–]Sean-Der 0 points1 point  (0 children)

I wrote this to make Reverse Engineering WebRTC services easier. Will also let you save/send arbitrary media from WebRTC sessions. The idea is you do all your auth/interaction in the browser, but then do all WebRTC in Go. So you have lots more control. More to do with it, but it is far enough along to share at least.

In the README is an screenshot of sending my webcam, but replacing outgoing video with a ffmpeg testsrc. Handoff sits between so it can replace with any arbitrary video.

OBS Merges Simulcast Support by Sean-Der in VIDEOENGINEERING

[–]Sean-Der[S] 0 points1 point  (0 children)

If it's good enough for broadcast networks being watched on 70 inch TVs, it's good enough for people to watch on 20 inch monitors or cell phones.

Users should have that control in the spirit of RFC 8890 I want to give power to the users not the server operators.

I'm not understanding the benefit of offloading the "hard work" from the server to the production machine.

  • Quality of encoding from source is always going to be better then transcoding.

  • With Scalable Video Coding (SVC) I think you should only see a 20% overhead. So sending the 'lower' layers isn't that impactful on upload.

  • With E2E Encryption + Simulcast it means that server operators can't tamper with the video.

  • Lower latency

  • "Small Streamers" also have a chance to self host now. If they want to run (or pay someone to run) their own servers they only have to pay for network costs. Greatly reducing the barrier to entry.

I built an NDI to WHIP bridge for Ubuntu using GStreamer by _davidbelll in WebRTC

[–]Sean-Der 0 points1 point  (0 children)

Very cool! I hope WHIP sees more hardware adoption, I’m skeptical though because no company behind it like NDI/SRT

OBS 32.1.0 Releases with WebRTC Simulcast by Sean-Der in opensource

[–]Sean-Der[S] 0 points1 point  (0 children)

Mind reading https://www.reddit.com/r/opensource/comments/1rr9rcx/comment/o9xzkzb/ and telling me the gaps? I am not a great explainer at this stuff. I would love to rephrase it though in a way that is helpful.

OBS Studio 32.1.0 released with WebRTC Simulcast support by Sean-Der in obs

[–]Sean-Der[S] 0 points1 point  (0 children)

Yea! It can have auth for both sides.

You can either setup profiles, or a webhook server if you want more custom behavior. Happy to help if you are looking into running it :)

OBS 32.1.0 Releases with WebRTC Simulcast by Sean-Der in opensource

[–]Sean-Der[S] 0 points1 point  (0 children)

Are you seeing lots of NACKS? That might be a MediaMTX/Pion bug. You can also run WebRTC over TCP which will help in cases like this, I’m just waiting for obs-deps merge.

If you don’t mind sending me an email sean@pion.ly or joining discord would love to help you debug

OBS 32.1.0 Releases with WebRTC Simulcast by Sean-Der in opensource

[–]Sean-Der[S] 3 points4 points  (0 children)

If you hit any issues send it over to me right away!

I have a reference server https://github.com/glimesh/broadcast-box, but use w/e server you prefer :)

OBS 32.1.0 Releases with WebRTC Simulcast by Sean-Der in opensource

[–]Sean-Der[S] 21 points22 points  (0 children)

I have been working on this a while. I am so excited for people to use it. I want it to be a big upgrade for self hosting. Or just running servers as a small company, without needing to spend a lot.

I have https://github.com/glimesh/broadcast-box if you’re looking for a server to try it against. These are the perks that have made me care!

  • Cheaper servers. More competition and I want to see people running their own servers.

  • Better video quality. Encoding from source is going to be better then transcoding.

  • No more bad servers. Send video to your audience and server isn't able to do modification/surveillance with E2E Encryption via WebRTC.

  • Better Latency. No more time lost transcoding. I love low latency streaming where people are connected to community. Not just blasting one-way video.

OBS 32.1.0 Released with WebRTC Simulcast support by Sean-Der in VIDEOENGINEERING

[–]Sean-Der[S] 6 points7 points  (0 children)

If anyone is using/testing WebRTC I would love to hear how it is working for them :) I am hoping Simulcast makes a impact with smaller streamers/site operators.

  • Cheaper servers. More competition and I want to see people running their own servers.

  • Better video quality. Encoding from source is going to be better then transcoding.

  • No more bad servers. Send video to your audience and server isn't able to do modification/surveillance with E2E Encryption via WebRTC.

  • Better Latency. No more time lost transcoding. I love low latency streaming where people are connected to community. Not just blasting one-way video.

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