> When the network is bad, you get... fewer JPEGs. That’s it. The ones that arrive are perfect.
This would make sense... if they were using UDP, but they are using TCP. All the JPEGs they send will get there eventually (unless the connection drops). JPEG does not fix your buffering and congestion control problems. What presumably happened here is the way they implemented their JPEG screenshots, they have some mechanism that minimizes the number of frames that are in-flight. This is not some inherent property of JPEG though.
> And the size! A 70% quality JPEG of a 1080p desktop is like 100-150KB. A single H.264 keyframe is 200-500KB. We’re sending LESS data per frame AND getting better reliability.
h.264 has better coding efficiency than JPEG. For a given target size, you should be able to get better quality from an h.264 IDR frame than a JPEG. There is no fixed size to an IDR frame.
Ultimately, the problem here is a lack of bandwidth estimation (apart from the sort of binary "good network"/"cafe mode" thing they ultimately implemented). To be fair, this is difficult to do and being stuck with TCP makes it a bit more difficult. Still, you can do an initial bandwidth probe and then look for increasing transmission latency as a sign that the network is congested. Back off your bitrate (and if needed reduce frame rate to maintain sufficient quality) until transmission latency starts to decrease again.
WebRTC will do this for you if you can use it, which actually suggests a different solution to this problem: use websockets for dumb corporate network firewall rules and just use WebRTC everything else
Probably either (1) they don't request another jpeg until they have the previous one on-screen (so everything is completely serialized and there are no frames "in-flight" ever) (2) they're doing a fresh GET for each and getting a new connection anyway (unless that kind of thing is pipelined these days? in which case it still falls back to (1) above.)
You can still get this backpressure properly even if you're doing it push-style. The TCP socket will eventually fill up its buffer and start blocking your writes. When that happens, you stop encoding new frames until the socket is able to send again.
They might want to check out what VNC has been doing since 1998– keep the client-pull model, break the framebuffer up into tiles and, when client requests an update, perform a diff against last frame sent, composite the updated tiles client-side. (This is what VNC falls back to when it doesn’t have damage-tracking from the OS compositor)
This would really cut down on the bandwidth of static coding terminals where 90% of screen is just cursor flashing or small bits of text moving.
If they really wanted to be ambitious they could also detect scrolling and do an optimization client-side where it translates some of the existing areas (look up CopyRect command in VNC).
Yup. Go look into tigervnc if you want to see the source. But also you can just search for "tigervnc h.264" and you'll see extensive discussions between the devs on h.264 and integrating it into tiger. This is something that people spent a LOT of brainpower on.
> When the network is bad, you get... fewer JPEGs. That’s it. The ones that arrive are perfect.
You can have still have weird broken stallouts though.
I dunno, this article has some good problem solving but the biggest and mostly untouched issue is that they set the minimum h.264 bandwidth too high. H.264 can do a lot better than JPEG with a lot less bandwidth. But if you lock it at 40Mbps of course it's flaky. Try 1Mbps and iterate from there.
And going keyframe-only is the opposite of how you optimize video bandwidth.
For normal video I think that's a good rule of thumb.
For mostly-static content at 4fps you can cut a bunch more bitrate corners before it looks bad. (And 2-3 JPEGs per second won't even look good at 1Mbps.)
Proper rate control for such realtime streaming would also lower framerate and/or resolution to maintain the best quality and latency they can over dynamic network conditions and however little bandwidth they have. The fundamental issue is that they don't have this control loop at all, and are badly simulating it by polling JPEGs.
There are so many things that I would have done differently.
> We added a keyframes_only flag. We modified the video decoder to check FrameType::Idr. We set GOP to 60 (one keyframe per second at 60fps). We tested.
Why muck around with P-frames and keyframes? Just make your video 1fps.
> Now it’s 10Mbps of blocky garbage that’s still 30 seconds behind.
10 Mbps is way too much. I occasionally watch YouTube videos where someone writes code. I set my quality to 1080p to be comparable with the article and YouTube serves me the video at way less than 1Mbps. I did a quick napkin math for a random coding video and it was 0.6Mbps. It’s not blocky garbage at all.
This blog post smells of LLM, both in the language style and the muddled explanations / bad technical justifications. I wouldn't be surprised if their code is also vibe coded slop.
Having pair programmed over some truly awful and locked down connections before, dropped frames are infinitely better than blurred frames which make text unreadable whenever the mouse is moved. But 40mbps seems an awful lot for 1080p 60fps.
Temporal SVC (reduce framerate if bandwidth constrained) is pretty widely supported by now, right? Though maybe not for H.264, so it probably would have scaled nicely but only on Webrtc?
Many moons ago I was using this software which would screenshot every five seconds and give you a little time lapse and the end of the day. So you could see how you were spending your computer time.
My hard disk ended up filling up with tens of gigabytes of screenshots.
I lowered the quality. I lowered the resolution, but this only delayed the inevitable.
One day I was looking through the folder and I noticed well almost all the image data on almost all of these screenshots is identical.
What if I created some sort of algorithm which would allow me to preserve only the changes?
I spent embarrassingly long thinking about this before realizing that I had begun to reinvent video compression!
So I just wrote a ffmpeg one-liner and got like 98% disk usage reduction :)
Why is video streaming so difficult? We've been doing it for decades, why is there seemingly no FOSS library which let's me encode an arbitrary dynamic frame rate image stream in Rust and get HD data with delta encoding in a browser receiver? This is insanity.
> The fix was embarrassingly simple: once you fall back to screenshots, stay there until the user explicitly clicks to retry.
There is another recovery option:
- increase the JPEG framerate every couple seconds until the bandwidth consumption approaches the H264 stream bandwidth estimate
- keep track latency changes. If the client reports a stable latency range, and it is acceptable (<1s latency, <200ms variance?) and bandwidth use has reached 95% of H264 estimate, re-activate the stream
Given that text/code is what is being viewed, lower res and adaptive streaming (HLS) are not really viable solutions since they become unreadable at lower res.
If remote screen sharing is a core feature of the service, I think this is a reasonable next step for the product.
That said, IMO at a higher level if you know what you're streaming is human-readable text, it's better to send application data pipes to the stream rather than encoding screenspace videos. That does however require building bespoke decoders and client viewing if real time collaboration network clients don't already exist for the tools (but SSH and RTC code editors exist)
Thinks: this video[1] is the processed feed from the Huygens space probe landing on Saturn's moon Titan circa 2005. Relayed through the Cassini probe orbiting Saturn, 880 million miles from the Sun. At a total mission cost of 3.25 billion dollars. This is the sensor data, altitude, speed, spin, ultra violet, and hundreds of photos. (Read the description for what the audio is encoding, it's neat!)
Look at the end of the video, the photometry data count stops at "7996 kbytes received"(!)
> "Turns out, 40Mbps video streams don’t appreciate 200ms+ network latency. Who knew. “Just lower the bitrate,” you say. Great idea. Now it’s 10Mbps of blocky garbage"
Yeah, I'm thinking the same thing. Capture the text somehow and send that, and reconstruct it on the other end; and the best part is you only need to send each new character, not the whole screen, so it should be very small and lightning fast?
If you are ok with a second or so of latency then MPEG-DASH (standardized version of HTTP Live Streaming) is likely the best bet. You simply serve the video chunks over HTTP so it should be just as compatible as the JPEG solution used here but provide 60fps video rather than crappy jpegs.
The standard supports adaptive bit rate playback so you can provide both low quality and high quality videos and players can switch depending on bandwidth available.
Super interesting. Some time ago I wrote some code that breaks down a jpeg image into smaller frames of itself, then creates an h.264 video with the frames, outputting a smaller file than the original image
You can then extract the frames from the video and reconstruct the original jpeg
Additionally, instead of converting to video, you can use the smaller images of the original, to progressively load the bigger image, ie. when you get the first frame, you have a lower quality version of the whole image, then as you get more frames, the code progressively adds detail with the extra pixels contained in each frame
It was a fun project, but the extra compression doesn’t work for all images, and I also discovered how amazing jpeg is - you can get amazing compression just by changing the quality/size ratio parameter when creating a file
They're just streaming a video feed of an LLC running in a terminal? Why not stream the actual text? Or fetch it piecemeal over AJAX requests? They complain that corporate networks support only HTTPS and nothing else? Do they not understand what the first T stands for?
This was the most entertaining thing I read all day. Kudos.
I've had similar experiences in the past when trying to do remote desktop streaming for digital signage (which is not particularly demanding in bandwidth terms). Multicast streaming video was the most efficent, but annoying to decode when you dropped data. I now wonder how far I could have gone with JPEGs...
You apparently need video for the 45 seconds window you then get before preventing catastrophic things to happen. From TFA:
> You’re watching the AI type code from 45 seconds ago
>
> By the time you see a bug, the AI has already committed it to main
>
> Everything is terrible forever
Is this satire? I mean: if the solution for things to not be terrible forever consists in catching what an AI is doing in 45 seconds (!) before the AI commits to trunk, I'm sorry but you should seriously re-evaluate your life plans.
I have some experience with pushing video frames over TCP.
It appears that the writer has jumped to conclusions at every turn and it's usually the wrong one.
The reason that the simple "poll for jpeg" method works is that polling is actually a very crude congestion control mechanism. The sender only sends the next frame when the receiver has received the last frame and asks for more. The downside of this is that network latency affects the frame rate.
The frame rate issue with the polling method can be solved by sending multiple frame requests at a time, but only as many as will fit within one RTT, so the client needs to know the minimum RTT and the sender's maximum frame rate.
The RFB (VNC) protocol does this, by the way. Well, the thing about rtt_min and frame rate isn't in the spec though.
Now, I will not go though every wrong assumption, but as for this nonsense about P-frames and I-frames: With TCP, you only need one I-frame. The rest can be all P-frames. I don't understand how they came to the conclusion that sending only I-frames over TCP might help with their latency problem. Just turn off B-frames and you should be OK.
The actual problem with the latency was that they had frames piling up in buffers between the sender and the receiver. If you're pushing video frames over TCP, you need feedback. The server needs to know how fast it can send. Otherwise, you get pile-up and a bunch of latency. That's all there is to it.
The simplest, absolutely foolproof way to do this is to use TCP's own congestion control. Spin up a thread that does two things: encodes video frames and sends them out on the socket using a blocking send/write call. Set SO_SNDBUF on that socket to a value that's proportional to your maximum latency tolerance and the rough size of your video frames.
One final bit of advice: use ffmpeg (libavcodec, libavformat, etc). It's much simpler to actually understand what you're doing with that than some convoluted gstreamer pipeline.
WebSockets over TCP is probably always going to cause problems for streaming media.
WebRTC over UDP is one choice for lossy situations. Media over Quic might be another (is the future here?), and it might be more enterprise firewall friendly since HTTP3 is over Quic.
> The constraint that ruined everything: It has to work on enterprise networks.
> You know what enterprise networks love? HTTP. HTTPS. Port 443. That’s it. That’s the list.
That's not enough.
Corporate networks also love to MITM their own workstations and reinterpret http traffic. So, no WebSockets and no Server-Side Events either, because their corporate firewall is a piece of software no one in the world wants and everyone in the world hates, including its own developers. Thus it only supports a subset of HTTP/1.1 and sometimes it likes to change the content while keeping Content-Length intact.
And you have to work around that, because IT dept of the corporation will never lift restrictions.
Back when I had a job at a big old corporation, a significant part of my value to the company was that I knew how to bypass their shitty MITM thing that broke tons of stuff, including our own software that we wrote. So I could solve a lot of problems people had that otherwise seemed intractable because IT was not allowed to disable it, and they didn't even understand the myriad ways it was breaking things.
The corporate firewall debate came up when we considered websockets at a previous company. Everyone has parroted the same information for so long that it was just assumed that websockets and corporate firewalls were going to cause us huge problems.
We went with websockets anyway and it was fine. Almost no traffic to the no-websockets fallback path, and the traffic that did arrive appeared to be from users with intermittent internet connections (cellular providers, foreign countries with poor internet).
I'm 100% sure there are still corporate firewalls out there blocking or breaking websocket connections, but it's not nearly the same problem in 2025 as it was in 2015.
If your product absolute must, no exceptions, work perfectly in every possible corporate environment then a fallback is necessary if you use websockets. I don't think it's a hard rule that websockets must be avoided due to corporate firewalls any more, though.
I've had to switch from SSE to WebSockets to navigate a corporate network (the entire SSE would have to close before the user received any of the response).
Then we ran into a network where WebSockets were blocked, so we switched to streaming http.
No trouble with streaming http using a standard content-type yet.
> And you have to work around that, because IT dept of the corporation will never lift restrictions.
Unless the corporation is 100% in-office, I’d wager they do in fact make exceptions - otherwise they wouldn’t have a working videoconferencing system.
The challenge is getting corporate insiders to like your product enough to get it through the exception process (a total hassle) when the firewall’s restrictions mean you can’t deliver a decent demo.
Oh, they'll do that anyway, once they find the workaround (Oh... you can paste a credit card if you put periods instead of dashes! Oh... I have to save the file and do it from my phone! Oh... I'll upload it as a .txt file and change the extension on the server!)
It's purely illusory security, that doesn't protect anything but does levy a constant performance tax on nearly every task.
>Oh, they'll do that anyway, once they find the workaround ...
This is assuming the DLP service blocks the request, rather than doing something like logging it and reported to your manager and/or CIO.
>It's purely illusory security, that doesn't protect anything but does levy a constant performance tax on nearly every task.
Because you can't ask deepseek to extract some unstructured data for you? I'm not sure what the alternative is, just let everyone paste info into deepseek? If you found out that your data got leaked because some employee pasted some data into some random third party service, and that the company didn't have any policies/technological measures against it, would your response still be "yeah it's fine, it's purely illusory security"?
I don't think that's a good read if the post you're implying this at. I think a more charitable read would be something like "people break rules for convenience so if your security relies on nobody breaking rules then you don't have thorough security".
You and op can be right at the same time. You imply the rules probably help a lot even while imperfect. They imply that pretending rules alone are enough to be perfect is incomplete.
Posting stuff into Deepseek is banned. The corporate firewall is like putting a camera in your home because you may break the law. But, yeah, arguing against cameras in homes because people find dead angles where they can hide may not be the strongest argument.
Sometimes they have checkboxes to tick in some compliance document and they must run the software that let them tick those checkboxes, no exceptions, because those compliances allow the company to be on the market. Regulatory captures, etc.
Believe me, the average Fortune 500 CEO does not know or care what “SSL MITM” is, or whether passwords should contain symbols and be changed monthly, or what the difference is between ‘VPN’ and ‘Zero Trust’.
They delegate that stuff. To the corporate IT department.
This is where the problems come from. Auditors are definitely what ultimately causes IT departments to make dumb decisions.
For example, we got dinged on an audit because instead of using RSA4096, we used ed25519. I kid you not, their main complaint was there wasn't enough bits which meant it wasn't secure.
This is 100% it- the auditor is confirming the system is configured to a set of requirements, and those requirements are rarely in lockstep with actual best practices.
where else are you going to find customers that are so sticky it will take years for them to select another solution regardless of how crappy you are. that will staff teams to work around your failures. who, when faced with obvious evidence of the dysfunction of your product, will roundly blame themselves for not holding it properly. gaslight their own users. pay obscene amounts for support when all you provide is a voice mailbox that never gets emptied. will happily accept your estimate about the number of seats they need. when holding a retro about your failure will happily proclaim that there wasn't anything _they_ could have done, so case closed.
I think the general idea/flow of things is "numbers go up, until $bubble explodes, and we built up smaller things from the ground up, making numbers go up, bloating go up, until $bubble explodes..." and then repeat that forever. Seems to be the end result of capitalism.
If you wanna kill corporate IT, you have to kill capitalism first.
I’d say there’s nothing inherently capitalist about large and stupid bureaucracies (but I repeat myself) spending money in stupid ways. Military bureaucracies in capitalist countries do it. Military bureaucracies in socialist countries did it. Everything else in end-stage socialist countries did it too. I’m sorry, it’s not the capitalism—things’d be much easier if it were.
I don't believe that. I don't necessarily love capitalism (though I can't say I see very many realistic better alternatives either), but if HN is full of people who could do corporate IT better (read: sanely), then the conclusion is just that corporate IT is run by morons. Maybe that's because the corporate owners like morons, but nothing about capitalism inherently makes it so.
playing devil's advocate for a second, but corpIT is also working with morons as employees. most draconian rules used by corpIT have a basis in at least one real world example. whether that example happened directly by one of the morons they manage or passed along from corpIT lore, people have done some dumb ass things on corp networks.
Yes, and the problem in that picture is the belief (whichever level of the management hierarchy it comes from) that you can introduce technical impediments against every instance of stupidity one by one until morons are no longer able to stupid. Morons will always find a way to stupid, and most organizations push the impediments well past the point of diminishing returns.
We did something similar in one of the places I've worked at. We sent xy coordinates and pointer events from our frontend app to our backend/3d renderer and received JPEG frames back. All of that wrapped in protobuf messages and sent via WS connection. Surpassingly it kinda worked, not "60fps worked" though obviously.
Yes, this is unfortunately still the way and was very common back when iOS Safari did not allow embedded video.
For a fast start of the video, reverse the implementation: instead of downgrading from Websockets to polling when connection fails, you should upgrade from polling to Websockets when the network allows.
Socket.io was one of the first libraries that did that switching and had it wrong first, too. Learned the enterprise network behaviour and they switched the implementation.
This reminds me of the time we built a big angular3 codebase for a content platform. When we had to launch, the search engines were expecting content to be part of page html while we are calling APIs to fetch the content ( angular3 didn’t have server side rendering at that point)
So only plausible thing to do was pre-build html pages for content pages and let load angular’s JS take its time to load ( for ux functionality). It looked like page flickered when JS loads for the first time but we solved the search engine problem.
I love the style of this blog-post, you can really tell that Luke has been deep down in the rabbit hole, encountered the Balrog and lived to tell the tale.
I was blown away when I realized I could stream mjpeg from a raspberry pi camera with lower latency and less ceremony than everything I tried with webrtc and similar approaches.
A long time ago I was trying to get video multiplexing to work over mobile over 3G. We struggled with H264 which had broad enough hardware support but almost no tooling and software support on the phones we were targeting. Even with engineers from the phone manufacturer as liaison we struggled to get access to any kind or SDK etc. We ended up doing JPEG streaming instead, much like the article said. And it worked great but we discovered we were getting a fraction of the framerate reported in Flash players - the call to refresh the screen was async and the act of receiving and deciding the next frame staved the redraw so the phone spent more time receiving lots of frames but not showing them. Super annoying and I don’t think the project survived long enough for us to find a fix.
No mention of PNGs? I don’t usually go to jpegs first for screenshots of text. Did png have worse compression? Burn more cpu? I’m sure there are good reasons, but it seems like they’ve glossed over the obvious choice here.
PNGs are lossless so you can’t really dial up the compression. You can save space by reducing to 8-bit color (or grayscale!) but it’s basically the equivalent of raw pixels plus zlib.
JPEG is extremely efficient to [de/en]code on modern CPUs. You can get close to 1080p60 per core if you use a library that leverages SIMD.
I sometimes struggle with the pursuit of perfect codec efficiency when our networks have become this fast. You can employ half-assed compression and still not max out a 1gbps pipe. From Netflix & Google's perspective it totally makes sense, but unless you are building a streaming video platform with billions of customers I don't see the point.
Would HLS be an option? I publish my home security cameras via WebRTC, but I keep HLS as a escape for hotel/cafe WiFi situations (MediaMTX makes it easy to offer both).
You can run all WebRTC traffic over a single port. It’s a shame you spent so much time/were frustrated by ICE errors
That’s great you got something better and with less complexity! I do think people push ‘you need UDP and BWE’ a little too zealously. If you have a homogeneous set of clients stuff like RTMP/Websockets seems to serve people well
This is similar to what BrowserBox does for the same reasons outlined. Glad to see the control afforded by "ye olde ways" is recognized and widely appreciated.
A very stupid hack that can work to "fix" this could be to buffer the h264 stream at the data center using a proxy before sending it to the real client, etc.
Their h264 iframes were bigger than the jpegs because they told the h264 encoder to produce bigger images. If they had set it to produce images the same size as the jpegs it most likely would have resulted in higher quality.
Eh, there are a few easy things one can try. Make sure to use a non-ancient kernel on the sender side (to get the necessary features), then enable BBR and NOTSENT_LOWAT (https://blog.cloudflare.com/http-2-prioritization-with-nginx...) to avoid buffering more than what's in-flight and then start dropping websocket frames when the socket says it's full.
Also, with tighter integration with the h264 encoder loop one could tell it which frames weren't sent and account for that in pframe generation. But I guess that wasn't available with that stack.
“We didn’t have the expertise to build the thing we were building, got in way over our heads, and built a basic POC using legacy technology, which is fine.”
If you have latency detection already why not pause H.264 frames, then when ack comes just force a key frame and resume (perhaps with adjusted target bitrate)?
I'm confused, do people actually watch their agents code like it was a screen share? Why does the AI even mess with that, just send a diff over text? Is it getting a keyboard next?
This is the definition of over-engineering. I don't usually criticize ideas but this is so stupid my head hurts.
The LinkedIn slop tone, random bolding, miscopied Markdown tables makes me invoke: "please read the copy you worked on with AI"
smaller thing: many, many, moons ago, I did a lot of work with H.264. "A single H.264 keyframe is 200-500KB." is fantastical.
Can't prove it wrong because it will be correct given arbitrary dimensions and encoding settings, but, it's pretty hard to end up with.
Just pulled a couple 1080p's off YouTube, biggest I-frame is 150KB, median is 58KB (`ffprobe $FILE -show_frames -of compact -show_entries frame=pict_type,pkt_size | grep -i "|pict_type=I"`)
This is a beautiful cope. Every time technology rolls out something that works great 90% of the time for 90% of the people, those 10%s pile up big time in support and lost productivity. You need functional systems that fall back gracefully to 1994 if necessary.
I started the first ISP in my area. We had two T1s to Miami. When HD audio and the rudiments of video started to increase in popularity, I'd always tell our modem customers, "A few minutes of video is a lifetime of email. Remember how exciting email was?"
This would make sense... if they were using UDP, but they are using TCP. All the JPEGs they send will get there eventually (unless the connection drops). JPEG does not fix your buffering and congestion control problems. What presumably happened here is the way they implemented their JPEG screenshots, they have some mechanism that minimizes the number of frames that are in-flight. This is not some inherent property of JPEG though.
> And the size! A 70% quality JPEG of a 1080p desktop is like 100-150KB. A single H.264 keyframe is 200-500KB. We’re sending LESS data per frame AND getting better reliability.
h.264 has better coding efficiency than JPEG. For a given target size, you should be able to get better quality from an h.264 IDR frame than a JPEG. There is no fixed size to an IDR frame.
Ultimately, the problem here is a lack of bandwidth estimation (apart from the sort of binary "good network"/"cafe mode" thing they ultimately implemented). To be fair, this is difficult to do and being stuck with TCP makes it a bit more difficult. Still, you can do an initial bandwidth probe and then look for increasing transmission latency as a sign that the network is congested. Back off your bitrate (and if needed reduce frame rate to maintain sufficient quality) until transmission latency starts to decrease again.
WebRTC will do this for you if you can use it, which actually suggests a different solution to this problem: use websockets for dumb corporate network firewall rules and just use WebRTC everything else
The trick is to not buffer frames on the sender.
This would really cut down on the bandwidth of static coding terminals where 90% of screen is just cursor flashing or small bits of text moving.
If they really wanted to be ambitious they could also detect scrolling and do an optimization client-side where it translates some of the existing areas (look up CopyRect command in VNC).
You can have still have weird broken stallouts though.
I dunno, this article has some good problem solving but the biggest and mostly untouched issue is that they set the minimum h.264 bandwidth too high. H.264 can do a lot better than JPEG with a lot less bandwidth. But if you lock it at 40Mbps of course it's flaky. Try 1Mbps and iterate from there.
And going keyframe-only is the opposite of how you optimize video bandwidth.
From the article:
“Just lower the bitrate,” you say. Great idea. Now it’s 10Mbps of blocky garbage that’s still 30 seconds behind.
10Mbps is still way too high of a minimum. It's more than YouTube uses for full motion 4k.
And it would not be blocky garbage, it would still look a lot better than JPEG.
For mostly-static content at 4fps you can cut a bunch more bitrate corners before it looks bad. (And 2-3 JPEGs per second won't even look good at 1Mbps.)
Video players used to call it buffering, and resolving it was called buffering issues.
Players today can keep an eye on network quality while playing too, which is neat.
> We added a keyframes_only flag. We modified the video decoder to check FrameType::Idr. We set GOP to 60 (one keyframe per second at 60fps). We tested.
Why muck around with P-frames and keyframes? Just make your video 1fps.
> Now it’s 10Mbps of blocky garbage that’s still 30 seconds behind.
10 Mbps is way too much. I occasionally watch YouTube videos where someone writes code. I set my quality to 1080p to be comparable with the article and YouTube serves me the video at way less than 1Mbps. I did a quick napkin math for a random coding video and it was 0.6Mbps. It’s not blocky garbage at all.
Temporal SVC (reduce framerate if bandwidth constrained) is pretty widely supported by now, right? Though maybe not for H.264, so it probably would have scaled nicely but only on Webrtc?
My hard disk ended up filling up with tens of gigabytes of screenshots.
I lowered the quality. I lowered the resolution, but this only delayed the inevitable.
One day I was looking through the folder and I noticed well almost all the image data on almost all of these screenshots is identical.
What if I created some sort of algorithm which would allow me to preserve only the changes?
I spent embarrassingly long thinking about this before realizing that I had begun to reinvent video compression!
So I just wrote a ffmpeg one-liner and got like 98% disk usage reduction :)
https://developers.google.com/speed/webp/gallery1
Screenshot once per second. Works everywhere.
I’m still waiting for mobile screenshare api support, so I could quickly use it to show stuff from my phone to other phones with the QR link.
There is another recovery option:
- increase the JPEG framerate every couple seconds until the bandwidth consumption approaches the H264 stream bandwidth estimate
- keep track latency changes. If the client reports a stable latency range, and it is acceptable (<1s latency, <200ms variance?) and bandwidth use has reached 95% of H264 estimate, re-activate the stream
Given that text/code is what is being viewed, lower res and adaptive streaming (HLS) are not really viable solutions since they become unreadable at lower res.
If remote screen sharing is a core feature of the service, I think this is a reasonable next step for the product.
That said, IMO at a higher level if you know what you're streaming is human-readable text, it's better to send application data pipes to the stream rather than encoding screenspace videos. That does however require building bespoke decoders and client viewing if real time collaboration network clients don't already exist for the tools (but SSH and RTC code editors exist)
Thinks: why not send text instead of graphics, then? I'm sure it's more complicated than that...
Look at the end of the video, the photometry data count stops at "7996 kbytes received"(!)
> "Turns out, 40Mbps video streams don’t appreciate 200ms+ network latency. Who knew. “Just lower the bitrate,” you say. Great idea. Now it’s 10Mbps of blocky garbage"
Who could do anything useful with 10Mbps. :/
[1] https://en.wikipedia.org/wiki/File:Huygens_descent.ogv
The standard supports adaptive bit rate playback so you can provide both low quality and high quality videos and players can switch depending on bandwidth available.
You can then extract the frames from the video and reconstruct the original jpeg
Additionally, instead of converting to video, you can use the smaller images of the original, to progressively load the bigger image, ie. when you get the first frame, you have a lower quality version of the whole image, then as you get more frames, the code progressively adds detail with the extra pixels contained in each frame
It was a fun project, but the extra compression doesn’t work for all images, and I also discovered how amazing jpeg is - you can get amazing compression just by changing the quality/size ratio parameter when creating a file
https://en.wikipedia.org/wiki/Teleprinter
I've had similar experiences in the past when trying to do remote desktop streaming for digital signage (which is not particularly demanding in bandwidth terms). Multicast streaming video was the most efficent, but annoying to decode when you dropped data. I now wonder how far I could have gone with JPEGs...
Why not just send text? Why do you need video at all?
(Although the fact they decided to use Moonlight in an enterprise product makes me wonder if their product actually was vibe coded)
> You’re watching the AI type code from 45 seconds ago > > By the time you see a bug, the AI has already committed it to main > > Everything is terrible forever
Is this satire? I mean: if the solution for things to not be terrible forever consists in catching what an AI is doing in 45 seconds (!) before the AI commits to trunk, I'm sorry but you should seriously re-evaluate your life plans.
I believe the latter can be adjusted in codec settings.
It appears that the writer has jumped to conclusions at every turn and it's usually the wrong one.
The reason that the simple "poll for jpeg" method works is that polling is actually a very crude congestion control mechanism. The sender only sends the next frame when the receiver has received the last frame and asks for more. The downside of this is that network latency affects the frame rate.
The frame rate issue with the polling method can be solved by sending multiple frame requests at a time, but only as many as will fit within one RTT, so the client needs to know the minimum RTT and the sender's maximum frame rate.
The RFB (VNC) protocol does this, by the way. Well, the thing about rtt_min and frame rate isn't in the spec though.
Now, I will not go though every wrong assumption, but as for this nonsense about P-frames and I-frames: With TCP, you only need one I-frame. The rest can be all P-frames. I don't understand how they came to the conclusion that sending only I-frames over TCP might help with their latency problem. Just turn off B-frames and you should be OK.
The actual problem with the latency was that they had frames piling up in buffers between the sender and the receiver. If you're pushing video frames over TCP, you need feedback. The server needs to know how fast it can send. Otherwise, you get pile-up and a bunch of latency. That's all there is to it.
The simplest, absolutely foolproof way to do this is to use TCP's own congestion control. Spin up a thread that does two things: encodes video frames and sends them out on the socket using a blocking send/write call. Set SO_SNDBUF on that socket to a value that's proportional to your maximum latency tolerance and the rough size of your video frames.
One final bit of advice: use ffmpeg (libavcodec, libavformat, etc). It's much simpler to actually understand what you're doing with that than some convoluted gstreamer pipeline.
WebRTC over UDP is one choice for lossy situations. Media over Quic might be another (is the future here?), and it might be more enterprise firewall friendly since HTTP3 is over Quic.
> You know what enterprise networks love? HTTP. HTTPS. Port 443. That’s it. That’s the list.
That's not enough.
Corporate networks also love to MITM their own workstations and reinterpret http traffic. So, no WebSockets and no Server-Side Events either, because their corporate firewall is a piece of software no one in the world wants and everyone in the world hates, including its own developers. Thus it only supports a subset of HTTP/1.1 and sometimes it likes to change the content while keeping Content-Length intact.
And you have to work around that, because IT dept of the corporation will never lift restrictions.
I wish I was kidding.
The corporate firewall debate came up when we considered websockets at a previous company. Everyone has parroted the same information for so long that it was just assumed that websockets and corporate firewalls were going to cause us huge problems.
We went with websockets anyway and it was fine. Almost no traffic to the no-websockets fallback path, and the traffic that did arrive appeared to be from users with intermittent internet connections (cellular providers, foreign countries with poor internet).
I'm 100% sure there are still corporate firewalls out there blocking or breaking websocket connections, but it's not nearly the same problem in 2025 as it was in 2015.
If your product absolute must, no exceptions, work perfectly in every possible corporate environment then a fallback is necessary if you use websockets. I don't think it's a hard rule that websockets must be avoided due to corporate firewalls any more, though.
Then we ran into a network where WebSockets were blocked, so we switched to streaming http.
No trouble with streaming http using a standard content-type yet.
Unless the corporation is 100% in-office, I’d wager they do in fact make exceptions - otherwise they wouldn’t have a working videoconferencing system.
The challenge is getting corporate insiders to like your product enough to get it through the exception process (a total hassle) when the firewall’s restrictions mean you can’t deliver a decent demo.
Request lives for longer than 15 sec? Fuck you.
Request POSTs some JSON? Maybe fuck you just a little bit, when we find certain strings in the payload. We won't tell you which though.
Because otherwise people do dumb stuff like pasting proprietary designs or PII into deepseek
It's purely illusory security, that doesn't protect anything but does levy a constant performance tax on nearly every task.
This is assuming the DLP service blocks the request, rather than doing something like logging it and reported to your manager and/or CIO.
>It's purely illusory security, that doesn't protect anything but does levy a constant performance tax on nearly every task.
Because you can't ask deepseek to extract some unstructured data for you? I'm not sure what the alternative is, just let everyone paste info into deepseek? If you found out that your data got leaked because some employee pasted some data into some random third party service, and that the company didn't have any policies/technological measures against it, would your response still be "yeah it's fine, it's purely illusory security"?
You and op can be right at the same time. You imply the rules probably help a lot even while imperfect. They imply that pretending rules alone are enough to be perfect is incomplete.
I try to remember many environments once likely supported Flash.
Wherever Tech is a first class citizen and seat at the corporate table, it can be different.
They delegate that stuff. To the corporate IT department.
It's all CyberSecurity insurance compliance that in many cases deviates from security best practices.
For example, we got dinged on an audit because instead of using RSA4096, we used ed25519. I kid you not, their main complaint was there wasn't enough bits which meant it wasn't secure.
Auditors are snake oil salesman.
If you wanna kill corporate IT, you have to kill capitalism first.
playing devil's advocate for a second, but corpIT is also working with morons as employees. most draconian rules used by corpIT have a basis in at least one real world example. whether that example happened directly by one of the morons they manage or passed along from corpIT lore, people have done some dumb ass things on corp networks.
For a fast start of the video, reverse the implementation: instead of downgrading from Websockets to polling when connection fails, you should upgrade from polling to Websockets when the network allows.
Socket.io was one of the first libraries that did that switching and had it wrong first, too. Learned the enterprise network behaviour and they switched the implementation.
https://developers.google.com/speed/webp/docs/webp_study
ALSO - the blog author could simplify - you don't need any code at all at the web browser.
The <img> tag automatically does motion jpeg streaming.
https://caniuse.com/jpegxl
So only plausible thing to do was pre-build html pages for content pages and let load angular’s JS take its time to load ( for ux functionality). It looked like page flickered when JS loads for the first time but we solved the search engine problem.
I love the style of this blog-post, you can really tell that Luke has been deep down in the rabbit hole, encountered the Balrog and lived to tell the tale.
I wonder if they just tried restarting the stream at a lower bitrate once it got too delayed.
The talk about how the images looks more crisp at a lower FPS is just tuning that I guess they didn't bother with.
- https://news.ycombinator.com/item?id=9954870
- https://phoboslab.org/log/2015/07/play-gta-v-in-your-browser...
JPEG is extremely efficient to [de/en]code on modern CPUs. You can get close to 1080p60 per core if you use a library that leverages SIMD.
I sometimes struggle with the pursuit of perfect codec efficiency when our networks have become this fast. You can employ half-assed compression and still not max out a 1gbps pipe. From Netflix & Google's perspective it totally makes sense, but unless you are building a streaming video platform with billions of customers I don't see the point.
You can run all WebRTC traffic over a single port. It’s a shame you spent so much time/were frustrated by ICE errors
That’s great you got something better and with less complexity! I do think people push ‘you need UDP and BWE’ a little too zealously. If you have a homogeneous set of clients stuff like RTMP/Websockets seems to serve people well
Maybe because the basic frequency transform is 4x4 vs 8x8 for JPG?
Hmm they must be doing something wrong, they're not that usually heavy.
Good engineering: when you're not too proud to do the obvious, but sort of cheesy-sounding solution.
Bargaining.
> closes tab
Eh, there are a few easy things one can try. Make sure to use a non-ancient kernel on the sender side (to get the necessary features), then enable BBR and NOTSENT_LOWAT (https://blog.cloudflare.com/http-2-prioritization-with-nginx...) to avoid buffering more than what's in-flight and then start dropping websocket frames when the socket says it's full.
Also, with tighter integration with the h264 encoder loop one could tell it which frames weren't sent and account for that in pframe generation. But I guess that wasn't available with that stack.
This is the definition of over-engineering. I don't usually criticize ideas but this is so stupid my head hurts.
smaller thing: many, many, moons ago, I did a lot of work with H.264. "A single H.264 keyframe is 200-500KB." is fantastical.
Can't prove it wrong because it will be correct given arbitrary dimensions and encoding settings, but, it's pretty hard to end up with.
Just pulled a couple 1080p's off YouTube, biggest I-frame is 150KB, median is 58KB (`ffprobe $FILE -show_frames -of compact -show_entries frame=pict_type,pkt_size | grep -i "|pict_type=I"`)
I started the first ISP in my area. We had two T1s to Miami. When HD audio and the rudiments of video started to increase in popularity, I'd always tell our modem customers, "A few minutes of video is a lifetime of email. Remember how exciting email was?"
Or is it intra-only H.264?
I mean, none of this is especially new. It's an interesting trick though!