I wouldn't say I'm done evaluating it, and as a spare-time project, my NVR's needs are pretty simple at present.
But WebCodecs is just really straightforward. It's hard to find anything to complain about.
If you have an IP camera sitting around, you can run a quick WebSocket+WebCodecs example I threw together: <https://github.com/scottlamb/retina> (try `cargo run --package client webcodecs ...`). For one of my cameras, it gives me <160ms glass-to-glass latency, [1] with most of that being the IP camera's encoder. Because WebCodecs doesn't supply a particular jitter buffer implementation, you can just not have one at all if you want to prioritize liveness, and that's what my example does. A welcome change from using MSE.
Skipping the jitter buffer also made me realize with one of my cameras, I had a weird pattern where up to six frames would pile up in the decode queue until a key frame and then start over, which without a jitter buffer is hard to miss at 10 fps. It turns out that even though this camera's H.264 encoder never reorders frames, they hadn't bothered to say that in their VUI bitstream restrictions, so the decoder had to introduce additional latency just in case. I added some logic to "fix" the VUI and now its live stream is more responsive too. So the problem I had wasn't MSE's fault exactly, but MSE made it hard to understand because all the buffering was a black box.
What was the WebRTC bug, would love to help! I saw at work that FireFox doesn't properly implement [0] I wanted to go fix after FFmpeg + WHEP.
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.
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.
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.
I built a Open Source tool to help people debug/understand their live stream quality better. I would really appreciate people's feedback! https://streamsniff.com/
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 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.
Broadcast Box is a project that I have been maintaining for a few years, but feels even more relevant with streaming services locking things down more and more.
I originally built Broadcast Box so I can get WHIP support into OBS, but it outgrew that. I love this new generation of streaming that is more intimate. Instead of pushing video to a large audience you pull a group of friends into a stream and low latency makes you feel more connected.
Always looking for feedback on the architecture, docs, and onboarding experience! especially suggestions to make setup easier for new users.
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.
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