- syscall interfaces are a mess, the primitive APIs are too slow for regular sized packets (~1500 bytes), the overhead is too high. GSO helps but it’s a horrible API, and it’s been buggy even lately due to complexity and poor code standards.

- the syscall costs got even higher with spectre mitigation - and this story likely isn’t over. We need a replacement for the BSD sockets / POSIX APIs they’re terrible this decade. Yes, uring is fancy, but there’s a tutorial level API middle ground possible that should be safe and 10x less overhead without resorting to uring level complexity.

- system udp buffers are far too small by default - they’re much much smaller than their tcp siblings, essentially no one but experts have been using them, and experts just retune stuff.

- udp stack optimizations are possible (such as possible route lookup reuse without connect(2)), gso demonstrates this, though as noted above gso is highly fallible, quite expensive itself, and the design is wholly unnecessarily intricate for what we need, particularly as we want to do this safely from unprivileged userspace.

- several optimizations currently available only work at low/mid-scale, such as connect binding to (potentially) avoid route lookups / GSO only being applicable on a socket without high peer-competition (competing peers result in short offload chains due to single-peer constraints, eroding the overhead wins).

Despite all this, you can implement GSO and get substantial performance improvements, we (tailscale) have on Linux. There will be a need at some point for platforms to increase platform side buffer sizes for lower end systems, high load/concurrency, bdp and so on, but buffers and congestion control are a high complex and sometimes quite sensitive topic - nonetheless, when you have many applications doing this (presumed future state), there will be a need.

https://lore.kernel.org/all/CALCETrVJqj1JJmHJhMoZ3Fuj685Unf=...

It shouldn’t be particularly hard to port over the optimization that prevents this problem for SOCK_PACKET. I’ll get to it eventually (might be quite a while), but I only care about this because of Tailscale, and I don’t have a ton of bandwidth.

I don't think io_uring is as complex as its reputation suggests. I don't think we need a substantially simpler low-level API; I think we need more high-level APIs built on top of io_uring. (That will also help with portability: we need APIs that can be most efficiently implemented atop io_uring but that work on non-Linux systems.)

uring is extremely problematic to integrate into many common application / language runtimes and it has been demonstrably difficult to integrate into linux safely and correctly as well, with a continual stream of bugs, security and policy control issues.

in principle a shared memory queue is a reasonable basis for improving the IO cost between applications and IO stacks such as the network or filesystem stacks, but this isn't easy to do well, cf. uring bugs and binder bugs.

One, uring is not extremely problematic to integrate, as it can be chained into a conventional event loop if you want to, or can even be fit into a conventionally blocking design to get localized syscall benefits. That is, you do not need to convert to a fully uring event loop design, even if that would be superior - and it can usually be kept entirely within a (slightly modified) event loop abstraction. The reason it has not yet been implemented is just priority - most stuff isn't bottlenecked on IOPS.

Two, yes you could have e middle-ground. I assume the syscall overhead you call out is the need to send UDP packets one at a time through sendmsg/sendto, rather than doing one big write for several packets worth of data on TCP. An API that allowed you to provide a chain of messages, like sendmsg takes an iovec for data, is possible. But it's also possible to do this already as a tiny blocking wrapper around io_uring, saving you new syscalls.

Let's take downloading a 1MB jpeg image over QUIC and rendering it on the screen.

I would hope that can be done in about 100k CPU cycles and 20 syscalls, considering that all the jpeg decoding and rendering is going to be hardware accelerated. The decryption is also hardware accelerated.

Unfortunately, no network API allows that right now. The CPU needs to do a substantial amount of processing for every individual packet, in both userspace and kernel space, for receiving the packet and sending the ACK, and there is no 'bulk decrypt' non-blocking API.

Even the data path is troublesome - there should be a way for the data to go straight from the network card to the GPU, with the CPU not even touching it, but we're far from that.

1. A 1 MB file is at the very least 64 individually encrypted TLS records (16k max size) sent in sequence, possibly more. So decryption 64 times is the maximum amount of bulk work you can do - this is done to allow streaming verification and decryption in parallel with the download, whereas one big block would have you wait for the very last byte before any processing could start.

2. TLS is still userspace and decryption does not involve the kernel, and thus no syscalls. The benefits of kernel TLS largely focus on servers sending files straight from disk, bypassing userspace for the entire data processing path. This is not really relevant receive-side for something you are actively decoding.

3. JPEG is, to my knowledge, rarely hardware offloaded on desktop, so no syscalls there.

Now, the number of actual syscalls end up being dictated by the speed of the sender, and the tunable receive buffer size. The slower the sender, the more kernel roundtrips you end upo with, which allows you to amortize the processing over a longer period so everything is ready when the last packet is. For a fast enough sender with big enough receive buffers, this could be a single kernel roundtrip.

1. This is pretty clean and straightforward. 2. This is obviously what we need to decouple a bunch of things without the previous downsides.

What has made it so hard to integrate it into common language runtimes? Do you have examples of where there's been an irreconcilable "impedance mismatch"?

much more approachable, boats has written about challenges integrating in rust: https://without.boats/tags/io-uring/

in the most general form: you need a fairly "loose" memory model to integrate the "best" (performance wise) parts, and the "best" (ease of use/forward looking safety) way to integrate requires C library linkage. This is troublesome in most GC languages, and many managed runtimes. There's also the issue that uring being non-portable means that the things it suggests you must do (such as say pinning a buffer pool and making APIs like read not immediate caller allocates) requires a substantially separate API for this platform than for others, or at least substantial reworks over all the existing POSIX modeled APIs - thus back to what I said originally, we need a replacement for POSIX & BSD here, broadly applied.

Zero-copy APIs will necessarily be tricky to implement and use, especially on memory safe languages.

Seems like the author points out two things:

1. The lack of rust futures supporting manual cancellation. That doesn't seem like an inevitable choice by rust.

2. Sharing buffers with kernel mode. This is probably a bigger topic.

This has not been true for a long time. There was an early design mistake that made it quite prone to these, but that mistake has been fixed. Unfortunately, the reputational damage will stick around for a while.

For context, to the best of my knowledge the current approach of the Linux CNA is, in keeping with long-standing Linux security policy of "every single fix might be a security fix", to assign CVEs regardless of whether something has any security impact or not.

CVE numbers are just a way to ensure everyone is talking about the same bug. Not every security issue has a CVE, not every CVE is a security issue.

Often, a regular bug turns out years later to have been a security issue, or a security issue turns out to have no security impact at all.

If you want a central authority to tell you what to think, just use CVSS instead of the binary "does it have a CVE" metric.

> The mission of the CVE® Program is to identify, define, and catalog publicly disclosed cybersecurity vulnerabilities [emphasis mine].

In fact, CVE stands for "Common Vulnerabilities and Exposures", again showing that CVE == security issue.

It's of course true that just because your code has an unpatched CVE doesn't automatically mean that your system is vulnerable - other mitigations can be in place to protect it.

> The CVE list aspires to describe and name all publicly known facts about computer systems that could allow somebody to violate a reasonable security policy for that system

There's also a decision from the editorial board on this, which said:

> Discussions on the Editorial Board mailing list and during the CVE Review meetings indicate that there is no definition for a "vulnerability" that is acceptable to the entire community. At least two different definitions of vulnerability have arisen and been discussed. There appears to be a universally accepted, historically grounded, "core" definition which deals primarily with specific flaws that directly allow some compromise of the system (a "universal" definition). A broader definition includes problems that don't directly allow compromise, but could be an important component of a successful attack, and are a violation of some security policies (a "contingent" definition).

> In accordance with the original stated requirements for the CVE, the CVE should remain independent of multiple perspectives. Since the definition of "vulnerability" varies so widely depending on context and policy, the CVE should avoid imposing an overly restrictive perspective on the vulnerability definition itself.

For more details, see https://web.archive.org/web/20000526190637fw_/http://www.cve... and https://web.archive.org/web/20020617142755/http://cve.mitre....

Under this definition, any kernel bug that could lead to user-space software acting differently is a CVE. Similarly, all memory management bugs in the kernel justify a CVE, as they could be used as part of an exploit.

So yes, any CVE is supposed to be a security problem, and it has always been so. Maybe not for your specific system or for your specific security posture, but for someone's.

Extending this to any bugfix is a serious misunderstanding of what an "exposure" means, and it is a serious difference from other CNAs. Linux CNA-assigned CVEs just can't be taken as seriously as normal CNAs.

How do you figure that? The original definition of CVE is exactly the same as how Linux approaches it.

Sure, in recent years security consultants have been overloading CVE to mean something else, but that's something to fix, not to keep.

> The CVE list aspires to describe and name all publicly known facts about computer systems that could allow somebody to violate a reasonable security policy for that system

As well as:

> Discussions on the Editorial Board mailing list and during the CVE Review meetings indicate that there is no definition for a "vulnerability" that is acceptable to the entire community. At least two different definitions of vulnerability have arisen and been discussed. There appears to be a universally accepted, historically grounded, "core" definition which deals primarily with specific flaws that directly allow some compromise of the system (a "universal" definition). A broader definition includes problems that don't directly allow compromise, but could be an important component of a successful attack, and are a violation of some security policies (a "contingent" definition).

> In accordance with the original stated requirements for the CVE, the CVE should remain independent of multiple perspectives. Since the definition of "vulnerability" varies so widely depending on context and policy, the CVE should avoid imposing an overly restrictive perspective on the vulnerability definition itself.

Under this definition, any kernel bug that could lead to user-space software acting differently is a CVE. Similarly, all memory management bugs in the kernel justify a CVE, as they could be used as part of an exploit.

> with specific flaws that directly allow some compromise of the system

> important component of a successful attack, and are a violation of some security policies

All of these are talking about security issues, not "acting differently".

If the kernel returned random values from gettime, that'd lead to tls certificate validation not being reliable anymore. As result, any bug in gettime is certainly worthy of a CVE.

If the kernel shuffled filenames so they'd be returned backwards, apparmor and selinux profiles would break. As result, that'd be worthy of a CVE.

If the kernel has a memory corruption, use after free, use of uninitialized memory or refcounting issue, that's obviously a violation of security best practices and can be used as component in an exploit chain.

Can you now see how almost every kernel bug can and most certainly will be turned into a security issue at some point?

Because no system has been ever taken down by code that behaved different from what it was expected to do? Right? Like http desync attacks, sql escape bypasses, ... . Absolutely no security issue going to be caused by a very minor and by itself very secure difference in behavior.

Additionally, any security bug in the kernel itself, so any use after free, any refcounting bug, any use of uninitialized memory.

Can you now see why pretty much every kernel bug fulfills that definition?

They are right, by the way. When CVEs were used for things like Heartbleed they made sense - you could point to Heartbleed's CVE number and query various information systems about vulnerable systems. When every single possible security fix gets one, AND automated systems are checking the you've patched every single one or else you fail the audit (even ones completely irrelevant to the system, like RCE on an embedded device with no internet access) the system is not doing anything useful - it's deleting value from the world and must be repaired or destroyed.

You've doubtless heard Tony Hoare's "There are two ways to write code: write code so simple there are obviously no bugs in it, or write code so complex that there are no obvious bugs in it.". Linux is definitely in the latter category, it's now such a sprawling system that determining whether a bug "really" has security implications is no long a reasonable task compared to just fixing the bug.

The other reason is that Linux is so widely used that almost no assumption made to simplify that above task is definitely correct.

What you'd want is the ability to control the buffer for the "raw network side", so that asynchronous network IO can be performed without having to copy between a raw network buffer and buffers owned by the TLS library.

It also would really help if TLS libraries supported processing multiple TLS records in a batched fashion. Doing roundtrips between app <-> tls library <-> userspace network buffer <-> kernel <-> HW for every 16kB isn't exactly efficient.

- Performance

- Stability

- Convenience

- Security

This differs from eg. Windows because on Windows the stable interface to the OS is in user-space, not tied to the syscall boundary. This has resulted in unfortunate compromises in the design of various pieces of OS functionality.

Thankfully things like futex and io-uring have dropped the "convenience" constraint from the syscall itself and moved it into user-space. Convenience is still important, but it doesn't need to be a constraint at the lowest level, and shouldn't compromise the other ideals.

As per a sibling comment, 1500 is just for Ethernet (the default, jumbo frames being able to go to (at least) 9000). But the Internet is more than just Ethernet.

If you're on DSL, then RFC 2516 states that PPPoE's MTU is 1492 (and you probably want an MSS of 1452). The PPP, L2TP, and ATM AAL5 standards all have 16-bit length fields allowing for packets up 64k in length. GPON ONT MTU is 2000. The default MTU for LTE is 1428. If you're on an HPC cluster, there's a good chance you're using Infiniband, which goes to 4096.

What are size do you suggest everyone on the planet go to? Who exactly is going to get everyone to switch to the new value?

65536

> Who exactly is going to get everyone to switch to the new value?

The same people who got everyone to switch to IPv6. It's a missed opportunity that these migrations weren't done at the same time imho.

It'll take a few decades, sure, but that's how big migrations go. What's the alternative? Making no improvements at all, forever?

I have some bad news...

> What's the alternative? Making no improvements at all, forever?

No, sadly. The alternative is what the entire tech world has been doing for the past 15 years: shove "improvements" inside whatever crap we already have because nobody wants to replace the crap.

If IPv6 were made today, it would be tunneled inside an HTTP connection. All the new apps would adopt it, the legacy apps would be abandoned or have shims made, and the whole thing would be inefficient and buggy, but adopted. Since poking my head outside of the tech world and into the wider world, it turns out this is how most of the world works.

What you're suggesting here wouldn't work, wrapping all the addressing information inside HTTP which relies on IP for delivery does not work. It would be the equivalent of sealing all the addressing information for a letter you'd like to send inside the envelope.

Given that one of the primary goals of IPv6 was increased address space, how would putting IPv6 in an HTTP connection riding over IPv4 solve that?

Sometimes for ease of mind I end up clamping it to v6 minimum (1280) just in case .

"GSO gains performance by enabling upper layer applications to process a smaller number of large packets (e.g. MTU size of 64KB), instead of processing higher numbers of small packets (e.g. MTU size of 1500B), thus reducing per-packet overhead."

Generally today an Ethernet frame, which is the basic atomic unit of information over the wire, is limited to 1500 bytes (the MTU, or Maximum Transmission Unit).

If you want to send more - the IP layer allows for 64k bytes per IP packet - you need to split the IP packet into multiple (64k / 1500 plus some header overhead) frames. This is called segmentation.

Before GSO the kernel would do that which takes buffering and CPU time to assemble the frame headers. GSO moves this to the ethernet hardware, which is essentially doing the same thing only hardware accelerated and without taking up a CPU core.

Basically (hyper simple), the kernel can lump stuff together when working with the network interface, which cuts down on ultra slow hardware interactions.

Please introduce your initialisms, if it's not guaranteed that first result in a search will be correct.

Searching for GSO network gives you the correct answer in the first result. I'd consider that condition met.

A system call to get that ancillary data adds overhead that can be amortized by having the application cache it, so that's probably the right design, and if it could be combined with sending (so a new flavor of sendto(2)) that would be even better, and it all has to be uring-friendly.

And that scares me, because there's not a single tool that has this on its radar for malware detection/prevention.

And the kernel has no business providing this middle-layer API. Why should it? Let people grab whatever they need from the ecosystem. Networking should be like Vulkan: it should have a high-performance, flexible API at the systems level with being "easy to use" a non-goal --- and higher-level facilities on top.

That and nobody was able to get performant microkernels working at the time, so we ended up with everything in the monokernel.

If you do trust the client processes then it could be better to just have them read/write IP packets though.

Linux scheduling from packet-received to thread has control is not real-time, and if the CPUs are busy, may be rather slow. That's probably part of the bottleneck.

The embarrassing thing is that QUIC, even in Google's own benchmarks, only improved performance by about 10%. The added complexity probably isn't worth the trouble. However, it gave Google control of more of the stack, which may have been the real motivation.

For background, the mental model of what receiving network data looks like in userspace is almost completely backwards compared to how general-purpose kernel network receive actually works. User code thinks it allocates a buffer (per-socket or perhaps a fancier io_uring scheme), then receives packets into that buffer, then processes them.

The kernel is the other way around. The kernel allocates buffers and feeds pointers to those buffers to the NIC. The NIC receives packets and DMAs them into the buffers, then tells the kernel. But the NIC and the kernel have absolutely no concept of which socket those buffers belong to until after they are DMAed into the buffers. So the kernel cannot possibly map received packets to the actual recipient's memory. So instead, after identifying who owns a received packet, the kernel retroactively charges the recipient for the memory. This happens on a per-packet basis, it involves per-socket and cgroup accounting, and there is no support for having a socket "pre-allocate" this memory in advance of receiving a packet. So the accounting is gnarly, involves atomic operations, and seems quite unlikely to win any speed awards. On a very cursory inspection, the TCP code seemed better tuned, and it possibly also won by generally handling more bytes per operation.

Keep in mind that the kernel can't copy data to application memory synchronously -- the application memory might be paged out when a packet shows up. So instead the whole charging dance above happens immediately when a packet is received, and the data is copied later on.

For quite a long time, I've thought it would be nifty if there was a NIC that kept received data in its own RAM and then allowed it to be efficiently DMAed to application memory when the application was ready for it. In essence, a lot of the accounting and memory management logic could move out of the kernel into the NIC. I'm not aware of anyone doing this.

I wonder if we could do a more advanced version of receive-packet steering that sufficiently identifies packets as definitely for a given process and DMAs them directly to that process's pre-provided buffers for later notification? In particular, can we offload enough information to a smart NIC that it can identify where something should be DMAed to?

In practice though, you only have a limited amount of these buffers, and it causes complications if multiple processes need to consume the same multicast.

The cloud doesn't implement multicast, but that doesn't mean it doesn't get used by people that build non-Internet networks and applications.

I’ve never seen the accelerated steering stuff work well in practice, sadly. The code is messy, the diagnostics are basically nonexistent, and it’s not clear to me that many drivers support it well.

1. dedicate a NIC to the application;

2. and have the userland app open a packet socket against the NIC, to drink from its firehose through MMIO against the kernel's own NIC DMA buffer;

...all without involving the kernel TCP/IP (or in this case, UDP/IP) stack, and any of the accounting logic squirreled away in there?

(You can also throw in a BPF filter here, to drop everything except UDP packets with the expected specified ip:port — but if you're already doing more packet validation at the app level, you may as well just take the whole firehose of packets and validate them for being targeted at the app at the same time that they're validated for their L7 structure.)

A lot of high end NICs support moderately complex receive queue selection rules.

This isn't GPUDirect though, that is an actual product.

- 64 packets per syscall, which is enough data to amortize the syscall overhead - a single packet is not.

- UDP offload optionally lets you defer checksum computation, often offloading it to hardware.

- UDP offload lets you skip/reuse route lookups for subsequent packets in a bundle.

What UDP offload is no good for though, is large scale servers - the current APIs only work when the incoming packet chains neatly organize into batches per peer socket. If you have many thousands of active sockets you’ll stop having full bundles and the overhead starts sneaking back in. As I said in another thread, we really need a replacement for the BSD APIs here, they just don’t scale for modern hardware constraints and software needs - much too expensive per packet.

If I had a dollar for every firewall vendor who thought dropping TCP retransmissions or TCP Reset was a good idea...

Not sure if that's changed, but at the time it wasn't worth having to consider rolling our own LBs.

To answer your original question, no, I haven't (knowingly) seen it on any public APIs.

Handling ACK packets in kernelspace would be one thing - helping for example RTT estimation. With userspace stack ACK's are handled in application and are subject to scheduler, suffering a lot on a loaded system.

What is needed is for QUIC offload to be invented/supported by HW so that most of the high-frequency/tiny-packet processing happens there, just as it does today for TCP offload. TCP large-send and large-receive offload is what is responsible for all the CPU savings as the application deals in 64KB or larger send/receives and the segmentation and receive coalescing all happen in hardware before an interrupt is even generated to involve the kernel, let alone userland.

Linux has receive packet steering, which can help with getting packets from the network card to the right CPU and the right userspace thread without moving from one CPU's cache to another.

You mean Receive Flow Steering, and RFS can only control RPS, so to do it in hardware you actually mean Accelerated RFS (which requires a pretty fancy NIC these days).

Even ignoring the hardware requirement, unfortunately it's not that simple. I find results vary wildly whether you should put process and softirq on the same CPU core (sharing L1 and L2) or just on the same CPU socket (sharing L3 but don't constantly blow out L1/L2).

Eric Dumazet said years ago at a Netdev.conf that L1 cache sizes have really not kept up with reality. That matches my experience.

QUIC doing so much in userspace adds another class of application which has a so-far uncommon design pattern.

I don't think it's possible to say whether any QUIC application benefits from RFS or not.

The Linux maintainers didn't want to be responsible for congestion collapse, but UDP lets you spray packets from userspace, so Google went with that.

https://www.fastly.com/blog/measuring-quic-vs-tcp-computatio... has good details and was done almost five years ago.

The solution isn't in more UDP offload optimizations as there aren't any semantics in UDP that are expensive other than the quantity and frequency of datagrams to be processed in the context of the QUIC protocol that uses UDP as a transport. QUIC's state machine needs to see every UDP datagram carrying QUIC protocol messages in order to move forward. Just like was done for TCP offload more than twenty years ago, portions of QUIC state need to move and be maintained in hardware to prevent the host from having to see so many high-frequency tiny state updates messages.

Even those garbage tg3 things from 1999 that OEMs are still putting onboard of enterprise servers have some TCP offload capability.

I think a kernel implementation of QUIC is the next logical step. A context switch to decrypt a packet header and send control traffic is just dumb. That's the kernel's job.

Userspace network stacks have never been a good idea. QUIC is no different.

(edit: Xin Long already has started a kernel implementation, see elsewhere on this page)

If you want to build a better TCP, do it. But hacking one in on top of UDP was a cheat that didn’t pay off. Well, assuming performance was even the actual goal.

I don't know how familiar the developers of QUIC were with SCTP in particular but they were definitely aware of the problems that prevented a better TCP from existing. The only practical solution is to build something on top of UDP, but if even that option proves unworkable, then the only other possibility left is to fragment the Internet.

If you've followed Dave Taht's bufferbloat stuff, the reason he lost faith in TCP is because middle devices have access to the TCP header and can interfere with it.

If SCTP got popular, then middle devices would ruin SCTP in the same way.

QUIC is the bufferbloat preferred solution because the header is encrypted. It's not possible for a middle device to interfere with QUIC. Endpoints, and only endpoints, control their own traffic.

I suspect part of the problem is that some of the rush is that people at major companies will get a promotion if they do "high impact" work out in the open.

HTTP/2 "solves head of line blocking" which is doesn't. It exchanged an HTTP SSL blocking issues with TCP on the real internet issue. This was predicted at the time.

The other issue is that instead of keeping it a simple protocol, the temptation to add complexity to aid a specific use case gets too much. (It's human nature I don't blame them)

H/1 pipelining was unusable, so H/1 had to wait for a response before sending the next request, which added a ton of latency, and made server-side processing serial and latency-sensitive. The solution to this was to open a dozen separate H/1 connections, but that multiplied setup cost, and made congestion control worse across many connections.

Indeed! and it works well on low latency, low packet loss networks. On high packet loss networks, it performs worse than HTTP1.1. Moreover it gets increasingly worse the larger the page the request is serving.

We pointed this out at the time, but were told that we didn't understand the web.

> H/1 pipelining was unusable,

Yup, but think how easy it would be to create http1.2 with better spec for pipe-lining. (but then why not make changes to other bits as well, soon we get HTTP2!) But of course pipelining only really works in a low packet loss network, because you get head of line blocking.

> open a dozen separate H/1 connections, but that multiplied setup cost

Indeed, that SSL upgrade is a pain in the arse. But connections are cheap to keep open. So with persistent connections and pooling its possible to really nail down the latency.

Personally, I think the biggest problem with HTTP is that its a file access protocol, a state interchange protocol and an authentication system. I would tentatively suggest that we adopt websockets to do state (with some extra features like optional schema sharing {yes I know thats a bit of enanthema}) Make http4 a proper file sharing prototcol and have a third system for authentication token generation, sharing and validation.

However the real world says that'll never work. So connection pooling over TCP with quick start TLS would be my way forward.

HTTP is a state interchange protocol. It's not any of the other things you mention.

"HTTP is being used as a file access, state interchange and authentication transport system"

Ideally we would split them out into a dedicated file access, generic state pipe (ie websockets) and some sort of well documented, easy to understand, implement and secure authentication mechanism (how hard can that be!?)

but to you point. HTTP was always mean to be stateless. You issue a GET request to find an object at a URI. That object was envisaged to be a file. (at least in HTTP 1.0 days) Only with the rise of CGI-bin in the middle 90s did that meaningfully change.

However I'm willing to bet that most of the traffic over HTTP is still files. Hence the assertion.

AJAX is a decent example. Microsoft's Outlook Web Access team implemented XMLHTTP as an activex thing for IE 5 and soon the rest of the vendors adopted it as a standard thing as XmlHttpRequest objects.

In fact, I suspect the list of things that exist in browsers because one vendor thought it was a good idea and everyone hopped on board is far, far longer than those designed by committee. Often times, the initially released version is not exactly the same that everyone standardized on, but they all get to build on the real-world consequences of it.

I happen to like the TC39 process https://tc39.es/process-document/ which requires two live implementations with use in the wild for something to get into the final stage and become an official part of the specification. It is obviously harder for something like a network stack than a JavaScript engine to get real world use and feedback, but it has helped to keep a lot of the crazier vendor specific features at bay.

>Next, we investigate more realistic scenarios by conducting the same file download experiments on major browsers: Chrome, Edge, Firefox, and Opera. We observe that the performance gap is even larger than that in the cURL and quic_client experiments: on Chrome, QUIC begins to fall behind when the bandwidth exceeds ∼500 Mbps.

Okay, well, this isn't going to be a problem over the general Internet, it's more of a problem in local networks.

For people that have high-speed connections, how often are you getting >500Mbps from a single source?

Often enough over HTTP/1.1 that discussions like this are relevant to my concerns.

I find this to be the issue when it comes to Google, and I bet it was known before hand; pushing processing to the user. For example, the AV1 video codec was deployed when no consumer had HW decoding capabilities. It saved them on space at the expense of increased CPU usage for the end-user.

I don't know what the motive was there; it would still show that they are carbon-neutral while billions are busy processing the data.

This was a bug. An improved software decoder was deployed for Android and for buggy reasons the YouTube app used it instead of a hardware accelerated implementation. It was fixed.

Having worked on a similar space (compression formats for app downloads) I can assure you that all factors are accounted for with decisions like this, we were profiling device thermals for different compression formats. Setting aside bugs, the teams behind things like this are taking wide-reaching views of the ecosystem when making these decisions, and at scale, client concerns almost always outweigh server concerns.

Also I will say if said overhead is not too much it's not that bad of a thing.

Moving interactions to JSON in many cases is just a better experience. If you click a Like button on Facebook, which is the better outcome: To see a little animation where the button updates, or for the page to reload with a flash of white, throw away the comment you were part-way through writing, and then scroll you back to the top of the page?

There's a reason XMLHttpRequest took the world by storm. More than that, jQuery is still used on more than 80% of websites due in large part to its legacy of making this process easier and cross-browser.

I don't understand how web devs understand the concept of loading and manipulating JSON to dynamically modify the page's HTML, but they don't understand the concept of loading and manipulating HTML to dynamically modify the page's HTML.

It's the same thing, except now you don't have to do a conversion from JSON->HTML.

There's no rule anywhere saying receiving HTML on the client should do a full page reload and throw up the current running javascript.

> XMLHttpRequest

This could've easily been HTMLHttpRequest and it would've been the same API, but probably better. Unfortunately, during that time period Microsoft was obsessed with XML. Like... obsessed obsessed.

Funny that you mention jQuery. When jQuery was hugely popular, people used it to make XMLHttpRequests that returned HTML which you then set as the innerHTML of some element. Of course being jQuery, people used the shorthand of `$("selector").html(...)` instead.

In the heyday of jQuery the JSON.parse API didn't exist.

arXiv typically contains pre-prints of papers. These may not have been peer-reviewed, and the contents may not reflect the actual "published" paper that was accepted (and/or corrected after peer review) to a conference or journal.

arXiv applies a watermark to the submitted PDF such that different versions are distinguishable on download.

As someone frequently at 150ms+ latency for a lot of websites (and semi-frequently 300ms+ for non-geo-distributed websites), in practice with the latency QUIC is easily the best for throughput, HTTP/1.1 with a decent number of parallel connections is a not-that-distant second, and in a remote third is HTTP/2 due to head-of-line-blocking issues if/when a packet goes missing.

I don't mind too much as long as they never try to take HTTP/1.1 from me.

Uptake on dekstop/laptops/servers is still extremely low and will be for a long time to come.

We were once very early with internet stuff, but now we lagging it seems.

Sadly here in Canada I don't think any ISP even supports IPv6 in any shape or form except for mobile. Videotron has been talking about it for a decade (and they have a completely outdated infrastructure now, only DOCSIS and a very bad implementation of it too), and Bell has fiber but does not provide any info on that either.

And if you click on the map view you will see "little corner of the woods" is ... the entire continent of Africa, huge countries like China and Indonesia.

Video streaming, sure, but we're already able to stream 4K video over a 25Mbit line. With modern internet connections being 200Mbit to 1Gbit, I don't see that we need the bandwidth in private homes. Maybe for video conferencing in large companies, but that also doesn't need to be 4K.

The underlying internet protocols are old, so there's no harm in assessing if they've outlived their usefulness. However, we should also consider in web applications and "always connected" is truly the best solution for our day to day application needs.

Private connections tend to be asymmetrical. In some cases, e.g. old DOCSIS versions, that used to be due to technical necessity.

Private connections tend to be unstable, the bandwidth fluctuates quite a bit. Depending on country, the actually guaranteed bandwidth is somewhere between half of what's on the sticker, to nothing at all.

Private connections are usually used by families, with multiple people using it at the same time. In recent years, you might have 3+ family members in a video call at the same time.

So if you're paying for a 1000/50 line (as is common with DOCSIS deployments), what you're actually getting is usually a 400/20 line that sometimes achieves more. And those 20Mbps upload are now split between multiple people.

At the same time, you're absolutely right – Gigabit is enough for most people. Download speeds are enough for quite a while. We should instead be increasing upload speeds and deploying FTTH and IPv6 everywhere to reduce the latency.

This bit:

    > IPv6 everywhere to reduce the latency

Google has measured to most customers about 20ms less latency on IPv6 than on IPv4, according to their IPv6 report.

I've run that comparison across four ISPs and never seen any significant difference in latency... not once in the decades I've had "dual stack" service.

I imagine that Google is getting confounded by folks with godawful middle/"security"ware that is too stupid to know how to handle IPv6 traffic and just passes it through.

    > but that also doesn't need to be 4K.

Zero trolling: Can you help me to better understand your last sentence?

    > However, we should also consider in web applications and "always connected" is truly the best solution for our day to day application needs.

I think it is bad design to expect a working internet connection, because in many places your can't expect bandwidth be cheap, or the connection to be stable. That's not to say that something like Google Docs (others seems to like it, but everyone in my company thinks it's awful) should be a thing, there's certainly value in the real time collaboration features, but it should be able to function without an internet connection.

Last week someone was complaining about the S3 (sleep) feature on laptops, and one thing that came to my mind is that despite these being portable, we somehow expect them to be always connected to the internet. That just seems like a somewhat broken mindset to me.

They are testing at 1gbps.

This is not regular 4g speed for sure, and it's a rare 5g speed. regular 5g speed is (in the US) 40-50mbps, so, 20x slower than they are testing.

I have 5gbps internet at home myself.

But that is not what i was replying to. I was replying to the claim that this affects regular 4g/5g cell phone speeds. The data is clear that it does not.

As said, i was only replying to the claim that this affects things at 4g/5g cell phone speeds, which it clearly does not, by their own data.

At the very least you probably would have to assign QUIC its own transport, rather than using UDP as "we have raw sockets at home". Problem is, only TCP and UDP reliably traverse the Internet[1]. Everything in the middle is sniffing traffic, messing with options, etc. In fact, Google rejected an alternate transport protocol called SCTP (which does all the stream multiplexing over a single connection that QUIC does) specifically because, among other things, SCTP's a transport protocol and middleboxes choke on it.

[0] I am aware that "SSL accelerators" used to do exactly this, but in modern times we have perfectly good crypto accelerators right in our CPU cores.

[1] ICMP sometimes traverses the internet, it's how ping works, but a lot of firewalls blackhole ICMP. Or at least they did before IPv6 made it practically mandatory to forward ICMP packets.

HTTP/3 implementations vary widely at the moment, and will likely take another decade to optimize to homogeneity. But even then, QUIC requires a lot of state management that TCP doesn't have to worry about (even in the kernel). There's a ton of processing involved with every UDP packet, and small MTUs, still engrained into many middle boxes and even end-user machines these days, don't make it any better.

So, yeah, as I felt about QUIC ... oh, about 6 years ago or so... HTTP/2 is actually really quite good enough for most use cases. The far reaches of the world and those without fast connections will benefit, but the majority of global transmissions will likely be best served with HTTP/2.

Intuitively, I consider each HTTP major version an increased order of magnitude in complexity. From 1 to 2 the main complexities are binary (that's debatable, since it's technically simpler from an encoding standpoint), compression, and streams; then with HTTP/3 there's _so, so much_ it does to make it work. It _can_ be faster -- that's proven -- but only when networks are slow.

TCP congestion control is its own worst enemy, but when networks aren't congested (and with the right algorithm)... guess what. It's fast! And the in-order packet transmissions (head-of-line blocking) makes endpoint code so much simpler and faster. It's no wonder TCP is faster these days when networks are fast.

I think servers should offer HTTP/3 but clients should be choosy when to use it, for the sake of their own experience/performance.

I see a few million daily page views: Memory usage has been down, latency has been down, network accounting (bandwidth) is about the same. Revenue (ads) is up.

> It _can_ be faster -- that's proven -- but only when networks are slow.

It can be faster in a situation that doesn't exist.

It sounds charitable to say something like "when networks are slow" -- but because everyone has had a slow network experience, they are going to think that QUIC would help them out, but real world slow network problems don't look like the ones that QUIC solves.

In the real world, QUIC wastes memory and money and increases latency on the average case. Maybe some Google engineers can come up with a clever heuristic involving TCP options or the RTT information to "switch on QUIC selectively" but honestly I wish they wouldn't bother, simply because I don't want to waste my time benchmarking another half-baked google fart.

I'm only interested in searches that advertisers are interested in, but this is also where Google gets their revenue from, so we are aligned with which users we want to prioritise, so I do not understand who you possibly think QUIC is for if not Google's ad business?

If google can't serve them an ad, they don't care about them, so QUIC isn't for them, so in order for QUIC to make business-sense to google it has to help them serve more ads.

The theory is if QUIC helped anyone out, Google (being huge) would be able to sell more ads to me if I use QUIC as well. In reality, they are able to sell me more ads when I turn off QUIC, which means this theory cannot be true.

Trying to make my sites load faster led me to experiment with QUIC and ultimately I didn't trust it enough to leave it on with the increase of complexity.

[0]: https://kiwee.eu/blog/http-3-how-it-performs-compared-to-htt...

HTTP2 I can't explain. HTTP2 should be better. I suspect it's probably an implementation bug because I can replicate lab performance and HTTP2 looks good to me in controlled tests.

I can always try turning it back on again in 6 months...

I mean, Lighthouse is maintained by Google (IIRC), and I can believe they are going to give their own protocol bonus points.

> Could this be down to server/client implementation of HTTP2 or would you say its a fundamental implication of the design of the protocol?

For stable internet connections, you'll see http2 beat http3 around 95% of the time. It's the 95th+ percentile that really benefits from http3 on a stable connection.

If you have unstable connections, then http3 will win, hands down.

Although I think for Intel CPUs the mmunuded to be disabled for years because their iGPU driver could not work with it. I hope things have improved with the Xe GPUs.

From my perspective, which is a web developer's, having QUIC, allowed the web standards to easily piggy back on top of it for the Webtransport API, which is ways better than the current HTTP stack and WebRTC which is a complete mess. Basically giving a TCP and UDP implementation for the web.

Knowing this, I feel like it makes more sense to me why Google choose this way of doing, which some people seem to be criticizing.

If you want your packets to reliably travel fairly unmolested between you and an effectively-randomly-chosen-peer on The Greater Internet, you have two transport protocol choices: TCP/IP or UDP/IP.

If you don't want the connection-management & etc that TCP/IP does for you, then you have exactly one choice.

> ...which some people seem to be criticizing.

People are criticizing the fact that on LAN link speeds (and fast (for the US) home internet speeds) QUIC is no better than (and sometimes worse than) previous HTTP transport protocols, despite the large amount of effort put into it.

It also seems that some folks are suggesting that Google could have put that time and effort into improving Linux's packet-handling code and (presumably) getting that into both Android and mainline Linux.

but as far as I can tell it's fast _enough_ just not as fast as it could be

mainly they seem to test situations related to bandwidth/latency which aren't very realistically for the majority of users (because most users don't have supper fast high bandwidth internet)

this doesn't meant QUIC can't be faster or we shouldn't look into reducing overhead, just it's likely not as much as a deal as it might initially loook

Out of curiosity: What's your use case to use ZScaler for this inspection instead?

So in essence, each browser tab or even each listening UDP socket could have a distinct IPv6 address dedicated to it, with packets delivered into a ring buffer in user-mode. This is so similar to what goes on with hypervisors now that existing hardware designs might even be able to handle it already.

Just an idle thought...

I think there may be a limit to how many IP addresses NICs (and maybe drivers) can track at once, though.

What I don't really get is why QUIC had to be invented when multi-stream protocols like SCTP already exist. SCTP brings the reliability of TCP with the multi-stream system that makes QUIC good for websites. Piping TLS over it is a bit of a pain (you don't want a separate handshake per stream), but surely there could be techniques to make it less painful (leveraging 0-RTT? Using session resumptions with tickets from the first connected stream?).

Secondly, the whole point of QUIC is to merge the TLS and transport handskakes into a single packet, to reduce RTT. This would mean you need to modify SCTP anyway to allow for this use case, so even what small support exists for SCTP in the large would need to be upgraded.

Thirdly, there is no reason to think that SCTP is better handled than UDP at the kernel's IP stack level. All of the problems of memory optimizations are likely to be much worse for SCTP than for UDP, as it's used far, far less.

TLS already does 0-RTT so you don't need QUIC for that.

The problem with UDP is that many optimisations are simply not possible. The "TCP but with blackjack and hookers" approach QUIC took makes it very difficult to accelerate.

SCTP is Fine™ on Linux but it's basically unimplemented on Windows. Acceleration beyond what these protocols can do right now requires either specific kernel/hardware QUIC parsing or kernel mode SCTP on Windows.

Getting Microsoft to actually implement SCTP would be a lot cleaner than to hack yet another protocol on top of UDP out of fear of the mighty shitty middleboxes.

The problem is not that Windows doesn't have in-kernel support for SCTP (there are several user-space libraries already available, you wouldn't even need to convince MS to do anything). The blocking issue is that many, many routers on the Internet, especially but not exclusively around all corporate networks, will drop any packet that is neither TCP or UDP over IP.

And if you think UDP is not optimized, I'd bet you'll find that the SCTP situation is far, far worse.

And regarding 0-RTT, that only works for resumed connections, and it is still actually 1 RTT (TCP connection establish). New connections still need 2-3 round trips (1 for TCP, 1 for TLS 1.3, or 2 for TLS 1.2) with TLS; they only need 1 round trip (even when using TLS 1.2 for encryption). With QUIC, you can have true 0-RTT traffic, sending the (encrypted) HTTP request data in the very first packet you send to a host [that you communicated with previously].

Of course, to open a raw socket you need privileged access, just like you do on Linux, because a raw socket allows you to see and respond to traffic from any other application (or even system traffic). But in principle you should be able to make a Service that handles SCTP traffic for you, and a non-privileged application could send its traffic to this service and receive data back.

I did find some user-space library that is purported to support SCTP on Windows [1], but it may be quite old and not supported. Not sure if there is any real interest in something like this.

[0] https://learn.microsoft.com/en-us/windows/win32/winsock/tcp-...

[1] https://www.sctp.de/sctp-download.html

when the latency/packet drop is low, prune the connections and you get monster speed.

When the latency/loss is high, grow the number of concurrent connections to overcome slow start.

Doesn't give you QUIC like multipath though.

If you have a valid argument to support your claim, why not present it?

You can read more about such things from, The Evolution of the Internet Congestion Control. https://groups.csail.mit.edu/ana/Publications/The_Evolution_...

A good solution is to create a newer protocol when the limits of an existing protcol are exceeded. No one thought of needing HTTPS long ago and now we have 443 for HTTP security. If we need something to be faster and it has already achieved an arbitrary limit for the sake of backward compatibility it would be better to introduce a new protocol.

I dislike the idea that we're turning into another Reddit where we are pointing fingers at people for updoots. If you dislike my opinion please present one equal to where that can be challenged.

It’s not clear to me how this is different to what’s happening. Is your objection that they did it on top of UDP instead of inventing a new transport layer?

Creating a new protocol starting from scratch would be better effort spent. A QUICv2 is on the way. https://datatracker.ietf.org/doc/rfc9369/

I don't think it addresses the problems with QUICv1 in terms of lightweight performance and bandwidth which the post claims QUIC lacks.

And if you have the new protocol available today, with excellent implementations for Linux, Windows, BSD, MacOS, Apple iOS, and for F5, Cisco, etc routers done, it will still take an absolute minimum of 5-10 years until it starts becoming available on the wider Internet, and that is if people are desperate to adopt it. And the vast majority of the world will not use it for the next 20 years.

The time for updating hardware to allow and use new protocols is going to be a massive hurdle to anything like this. And the advantage to doing so over just using UDP would have to be monumental to justify such an effort.

The reality is that there will simply not be a new transport protocol used on the wide internet in our lifetimes. Trying to get one to happen is a pipe dream. Any attempts at replacing TCP will just use UDP.

However, in reality, the people/forces pushing for HTTP2 and QUIC are the same one(s?) who have a defacto monopoly on browsers.

So, yes, it's a political issue, and they just implemented their changes on a layer (or even... "app") that they had the most control over.

On a purely "moral" perspective, political expediency probably shouldn't be the reason why something is done, but of course that's what actually happens in the real world...

The problems with SCTP, or any new transport-layer protocol (or any even lower layer protocol), run much deeper than deploying a new protocol on any higher layer.

“Its purpose is to combat various ossification vectors and exercise the version negotiation framework.”

I initially understood your first post as: "Let's not try to make the internet faster"

With this reply, you are clarifying your initial post that was very unclear. Now I understand it as:

"Let's not try to make existing protocols faster, let's make new protocols instead"

In fact our modern network infrastructure returns on designs intended for limited network performance. Our networks are fiber and 5g which are roughly 170,000 times faster and wider since the initial inception of the internet.

Time for a QUICv2

https://datatracker.ietf.org/doc/rfc9369/

But I don't think it addresses the disparity between it and lightweight protocols as networks get faster.

I think that GoogleHTTP has real-world uses for bad connectivity or in datacenters where they can fine-tune their data throughput (and buy crazy good NICs), but it seems that to use it for replacing TCP (which seems to be confirmed as very good when receiver and sender aren't controlled by the same party) the world needs a hardware overhaul or something.