Comparing HTTP/3 vs. HTTP/2 Performance(blog.cloudflare.com) |
Comparing HTTP/3 vs. HTTP/2 Performance(blog.cloudflare.com) |
> For a small test page of 15KB, HTTP/3 takes an average of 443ms to load compared to 458ms for HTTP/2. However, once we increase the page size to 1MB that advantage disappears: HTTP/3 is just slightly slower than HTTP/2 on our network today, taking 2.33s to load versus 2.30s
And in their closer to real world benchmarks, they find no improvement, instead some negligible degradation.
> As you can see, HTTP/3 performance still trails HTTP/2 performance, by about 1-4% on average in North America and similar results are seen in Europe, Asia and South America. We suspect this could be due to the difference in congestion algorithms: HTTP/2 on BBR v1 vs. HTTP/3 on CUBIC. In the future, we’ll work to support the same congestion algorithm on both to get a more accurate apples-to-apples comparison.
As a developer of web apps, I will personally continue to not think that much about HTTP/3. Perhaps in the future network/systems engineers will have figured out how to make it bear fruit? I don't know, but it seems to me of unclear wisdom to count on it.
In TCP, a connection is uniquely identified by the following tuple:
(src ip, src port, dst ip, dst port)
We can do some trickery to let people keep their addresses while inside of a network, but switch from mobile data to wifi and every TCP connection drops.
This is easy enough to solve, in theory. Give every connection an unique ID, and then remember the last address you received a packet for that connection from, ideally in the kernel. This makes IP addresses completely transparent to applications, just like MAC addresses are. However, the tuple is assumed almost everywhere and NAT makes new layer 4 protocols impossible. Unless you layer them over UDP. And this is exactly what Wireguard, QUIC, mosh and others do. Once it's ubiquitous, you'll be able to start an upload or download at home, hop on your bike, ride to the office, and finish it without the connection dropping once.
...and part of me think HTTP/3 could be that universal transport protocol that could eventually solve this problem, and then I agree.
you could do this for over 20 years as long as server supports HTTP/1.1 https://en.wikipedia.org/wiki/Byte_serving
One of the things I was looking forward to from IPv6 was the mobility of IPs [1].
[1] - https://en.wikipedia.org/wiki/Mobile_IP#Changes_in_IPv6_for_...
Doesn't that mean we need to rethink TCP instead of pulling all these transfer concerns to the application layer?
Chrome is happily using QUIC, also on Android. And for non-browser apps, every Android app that links against the recommended network client library will do QUIC transparently.
https://developer.android.com/guide/topics/connectivity/cron...
> On a well-optimized site like Google Search, connections are often pre-established, so QUIC’s faster connections can only speed up some requests—but QUIC still improves mean page load time by 8% globally, and up to 13% in regions where latency is higher.
https://cloudplatform.googleblog.com/2018/06/Introducing-QUI...
Another summary: https://kinsta.com/blog/http3/
Older Cloudflare blog: https://blog.cloudflare.com/http3-the-past-present-and-futur...
Personally, whenever I've experienced "this site is slow", it's either because the server is really congested or there's something else taking the time (like copious amounts of JS executing on the client); both cases in which the tiny improvements (if any) of a protocol version would have zero effect.
When you consider that there's an additional huge chunk of complexity (= bugs) added by these new protocols, for small or even negative performance improvement, it really seems like there is no value being added --- it's more of a burden on everyone except those working on it.
Syncthing uses a bunch of protocols by default to find clients, including TCP, QUIC and a few others. If you want to have some insight into how your network behaves with these protocols head-to-head, spin up syncthing and wireshark.
> With HTTP/2, any interruption (packet loss) in the TCP connection blocks all streams (Head of line blocking). Because HTTP/3 is UDP-based, if a packet gets dropped that only interrupts that one stream, not all of them.
So while HTTP/3 on a perfect network might be a 1 - 4% slower it's more stable/reliable in that any packet loss won't cause a dramatic drop off in performance... so 1-4% in best case network conditions but in real world network conditions http/3 should~ be much better...
Congestion control algorithm - and congestion window sizing/tuning - plays a not-insignificant role in throughput, especially when comparing a 15KiB object vs. a 1MiB object. It's often _more_ outsized for those "medium sized" objects, as too small a window won't scale up by the time the transfer completes in some cases.
In other words: this is a good post, but the caveats around congestion control algorithm are a little understated w.r.t. the benchmarks.
> Overall, we’re very excited to be allowed to help push this standard forward. Our implementation is holding up well, offering better performance in some cases and at worst similar to HTTP/2. As the standard finalizes, we’re looking forward to seeing browsers add support for HTTP/3 in mainstream versions.
I feel like the conclusion should be "Hypothetical advantages of HTTP/3 still not realized," but they are "We're excited to be working on this", with no mention of... why. Like, why isn't HTTP/3 resulting in expected advantages; what might be changed to change this; what are you going to do to try to realize actual advantages to 'push the standard forward'? It seems like a standard for the sake of saying you have done something and have a standard, if there aren't any realized advantages, no?
You could be midway through a gaming session over websocket, and walk away from your wifi, and you shouldn't notice a glitch.
Nearly nothing else offers that ability, and it's very annoying, especially in offices with hundreds of wifi access points - I should be able to walk down the corridor on a video call without glitchiness!
MPTCP (developed mostly by Apple) offers the same, but Google and Microsoft are holding it back, for some unknown reason.
I know of a standard for 802.1X preauthentication, that does 802.1X authentication via old AP before roaming.
If your APs aren't doing NAT or stateful firewall, then there is no state to transfer, except automatic updating of MAC addresses on switches, and the authentication of the client which shouldn't require any cross-communication unless using 802.1X with mentioned preauthentication. You will lose packets in flight while switching, but it shouldn't take long.
With a good network connection with little packet loss, I wouldn't expect much benefit to /3. Especially since all the server and client implementations are immature and in user space without kernel support.
The benefits should show up with (poor) mobile connections.
For me the most exciting part is the seamless network switching potential of /3 on mobile devices
This issue is really noticeable on my crappy home mobile internet when loading web pages, in combination with the timeout being absurdly long for reasons I don't understand.
I know there's a push to get software to support an offline mode, but I wish there was a similar push to improve software when in lie-fi.
Under firefox you can set "network.http.sdpy.enable" to false to switch back to HTTP/1.
The improvement I have with HTTP2 is hardly noticeable, but HOL blocking is very tangible as soon as you have occasional random packet loss.
Thanks! I might have to try that out, currently relegated to tunneling everything over sshuttle which of course makes things slower but reliable since it decompiles the TCP packets making them appear to work flawlessly to HTTP... HTTP1 may be a faster solution.
[edit]
Trying this now. Correction on parent comment for Firefox:
goto:
about:config
search:
network.http.spdy.enabled.http2Found an article explaining the concept, called selective acknowledgment:
https://en.wikipedia.org/wiki/Transmission_Control_Protocol#...
Would reducing the retransmission delay be sufficient? Or simply letting the browser open two connections to a standard HTTP port?
This is not true. The only HTTP/2 feature that doesn't fit into the traditional HTTP semantics is PUSH. And even that is the request/response model - the only difference is that the request is injected also from the server side and not being received from the client. We just pretend we would have received such a request from the client, send the response towards the client, and hope the client won't reject it.
When I compared late last year I found HTTP/3 to be noticeably slower, https://pgjones.dev/blog/early-look-at-http3-2019/ however my test was much less comprehensive than the one here.
The gist of it was that Quic tends to just flat out choke out TCP running on the same network paths?
Anyone know about this?
There is some mention of BBRv2 improving fairness but not the outside academic paper I was looking for -
https://datatracker.ietf.org/meeting/106/materials/slides-10...
This is a surprisingly difficult problem, especially given the constraints of using pure JS. Some issues that spring to mind included:
- The User-Agent is meaningless on iPhones, basically because Steve Jobs got sick of leaking new models in Apache logs. There are other ways of figuring this out but it's a huge pain.
- Send too much traffic and you can crash the browser, particularly on mobile devices;
- To maximize throughput it became necessary to use a range of ports and simultaneously communicate on all of them. This in turn could be an issue with firewalls;
- Run the test too long and performance in many cases would start to degrade;
- Send too much traffic and you could understate the connection speed;
- Sending larger blobs tended to be better for measuring throughput but too large could degrade performance or crash the browser. Of course, what "too large" was varied by device;
- HTTPS was abysmal for raw throughput on all but the beefiest of computers;
- To get the best results you needed to turn off a bunch of stuff like Nagel's algorithm and any implicit gzip compression;
- You'd have to send random data to avoid caching even with careful HTTP headers that should've disabled caching.
And so on.
Perhaps the most vexing issue that I was never able to pin down was with Chrome on Linux. In certain circumstances (and I never figured out what exactly they were other than high throughput), Chrome on Linux would write the blobs it downloaded to /tmp (default behaviour) and never release them until you refreshed the Webpage. And no there were no dangling references. The only clue this was happening was that Chrome would start spitting weird error messages to the console and those errors couldn't be trapped.
So pure JS could actually do a lot and I actually spent a fair amount of effort to get this to accurately show speeds up to 10G (I got up to 8.5G down and ~7G up on Chrome on a MBP).
But getting back to the article at hand, what you tend to find is how terribly TCP does with latency. A small increase in latency would have a devastating effect on reported speeds.
Anyone from Australia should be intimately familiar with this as it's clear (at least to me) that many if not most services are never tested on or designed for high-latency networks. 300ms RTT vs <80ms can be the difference between a relatively snappy SPA and something that is utterly unusable due to serial loads and excessive round trips.
So looking at this article, the first thing I searched for was the word "latency" and I didn't find it. Now sure the idea of a CDN like Cloudfare is to have a POP close to most customers but that just isn't always possible. Plus you hit things not in the CDN. Even DNS latency matters here where pople have shown meaningful improvements in Web performance by just having a hot cache of likely DNS lookups.
The degradation in throughput in TCP that comes from latency is well-known academically. It just doesn't seem to be known about, given attention to or otherwise catered for in user-facing services. Will HTTP/3 help with this? I have no idea. But I'd like to know before someone dismisses it as having minimal improvements or, worse, as degrading performance.
Will we see more performance tuning when it comes to MTU sizes?
Surprised to hear that. Sending data should never lead to a crash. Even an aborted request wouldn't be great. When was that? Hope these things got fixed.
Also 0-RTT requests still follow the request/response model.
QUIC handoffs are a lot more complicated. They will require a library which supports all the necessary features. And it will require infrastructure which supports it. Without infrastructure support, packets from the client might get routed to the wrong host after a IP tuple change, and can from there on not associated with the QUIC connection.
My guess is some QUIC deployments will figure out how to make it work - others likely won't, since a lot of efforts is involved.
However, the solution should not and cannot be a global routing table. Aside from privacy issues, routing tables are very expensive. It needs to be decentralized, done on or at least near the endpoints, similar to programs like mosh, wireguard etc. Though there are certain security and privacy trade-offs in mentioned protocols that may not be appropriate.
The "flat address space" idea however is completely ridiculous. That would mean every node on the internet keeping track of the path to every singe other node. This is what ethernet does and it barely scales to ten or so thousand nodes. Routers are already struggling hard with the 700k table entries we have for the IPv4 internet. To the point where providers actually wrap IP packets inside of simpler protocols once they enter the network.
We need some kind of hierarchical addressing that expresses the location of a node in the network. We know this works. We just need the layers above not to rely on those addresses staying constant.
If anything, we need a more hierarchical structure, with stricter separations by, e.g. Continent/Country/Province and possibly down to street or even house level, so routers can more easily just throw the data in the right general direction. Note the obvious privacy problem there.
Acknowledgements are encrypted.
You may also be interested in SCTP, which is awesome on paper and works well across the internet. But since most firewalls only understand TCP, UDP, and ICMP other protocols get auto dropped.
SCTP could have been amazing. https://en.m.wikipedia.org/wiki/Stream_Control_Transmission_...
It still is, it's part of the WebRTC spec, and when you use a WebRTC data channel, you're using SCTP over DTLS over UDP! (or TCP, possibly with a TURN relay, which may end up tunneling the whole thing over TLS over TCP :))
There are a lot of acronyms in WebRTC, thankfully there's https://webrtcglossary.com
https://orchid.com VPN does tunnel the traffic over webrtc.
On the other hand, there will never be a time when TCP could possibly be "update", it is too entrenched and fossilized.
I know it's possible for wireguard, maybe mosh.