I hope it’s getting better with education/more libraries. It’s also amazing how easy Codex etc… can burn through it now
4,712 karma · joined November 18, 2012
Socials: - github.com/Sean-Der - linkedin.com/in/sean-dubois - siobud.com - pion.ly
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I hope it’s getting better with education/more libraries. It’s also amazing how easy Codex etc… can burn through it now
However things like 'Call center helpline' turn based actually seems better! You don't want to be interrupted when giving information back and forth (I think?)
You can't beat Websockets :) Especially since you have so much tooling/existing stuff that works with HTTP.
I have been trying to get a website off the ground that does Datachannels + SQlite in the browser and then users sync between each other. I have gotten distracted so many times though.
I think It’s a case of you improve what you own. The owners of WebRTC servers were aggressively improving their part. They don’t own the inference servers.
Even for clients you have things like libpeer that libwebrtc can't hit.
Do any alternatives to TvHeadEnd exist? I looked a bit and seems like it’s the best. So much customization it would be hard to reach it’s quality
Would you mind explaining more? If you are doing WHIP/WHEP you should be able to drop in Broadcast Box/MediaMTX etc... and switch out servers and no one should notice. You can use browser/mobile/ffmpeg/OBS etc... get the same behavior. I care a lot about the broadcast space, want to learn about other problems.
> subtly speed up audio/video to keep everything in sync
You can use https://webrtc.googlesource.com/src/+/refs/heads/main/docs/n... to add more delay (if you want to force more buffering). Or if you don't link the media together (via MediaStream) you don't get the behavior you describe either!
> capture each participant's audio individually
That's a neat problem. I haven't solved this one myself, I wonder if it's easier with RtpTransport or insertable streams?
Never give up! Free/Open Source software especially in schools is so important
ex_webrtc is super cool. They have a cool built-in dashboard/analytics flow. It is way more 'operations friendly' then Pion it seems. I haven't used it heavily myself though.
You need to make sure you are sending the same codec that the remote expects, otherwise nothing else! You can do a different resolution, bitrate etc...
* Do video playback out of the browser. You can render a subset of frames, use a different pipeline for decode etc...
* Pull video from a different source. Join Google Meet on current computer, but stream from another host.
I am also using v4l2loopback, but its annoying to juggle /dev/video* devices. I wanted to do video stuff in docker containers, and it would be amazing if I could do pipewire in each container and have no global state.
I couldn't get anything to work in Chromium. FireFox saw the device, but video didn't come across.
I ended up doing proxy because Google Meet doesn’t let me hook at any RTCPeerConnection APIs at all. I wanted to send synthetic media in, but couldn’t get it working. Ending up doing a virtual webcam on Linux.
With ‘media-send’ I can send it out to ffmpeg/GStreamer and that does all the heavy lifting
No better feeling to work on something and hear it is novel! So many projects that I think will be useful miss the mark.
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.
Otherwise if everything is trusted I love that we can have conversations without depending on others over the internet :)
RTMP has SO many users for video (Twitch, YouTube etc...) yet you have librtmp which has so many forks. OBS has its own version in-tree....
Then at the same time you have people trying to add extensions to RTMP still!
RTMP has held back the use cases I have cared about since ~2015 so I am excited to see people embrace other options.
Just features/software need to be implemented?
If you are still struggling with WebRTC problems would love to help. Pion has a Discord and https://webrtcforthecurious.com helps a bit to understand the underlying stuff, makes it easier to debug.
[0] https://datatracker.ietf.org/doc/html/rfc8445#section-7.2.5....
Most days I’m more excited for this space than Pion. It’s just more fun to make software for users vs businesses mostly doing AI maybe?
* 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.
* 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.
I see a constant stream of people saying 'my video is blurry' or 'why do I have so much latency' and I wanted to build something that helped people understand/diagnose. Also wanted to make something that runs in the browser so you can share/send a link to a friend (or on discord) to make debugging easier.
I wanted to help people debug/understand their live stream quality better. I would really appreciate people's feedback!
I see a constant stream of people saying 'my video is blurry' or 'why do I have so much latency' and I wanted to build something that helped people understand/diagnose. Also wanted to make something that runs in the browser so you can share/send a link to a friend (or on discord) to make debugging easier.
I am hoping to move it to https://github.com/glimesh org soon so it can sit next to broadcast-box.
I also love seeing it used for 'kill the jump box' and file transfer. Just drives me crazy that we lets files sit on file providers.
Especially if you are transferring in the office! Send it right over the LAN and could be instant. Being forced to upload + download from remote servers frustrates me.