The Size of Packets
54 comments
·April 18, 2025hliyan
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kazinator
Another thing to look into in this kind of situation is enabling jumbo frames.
ay
“You had an MTU problem. You enable jumbo frames. Now you have two MTU problems”
Unless you control the entire set of possible paths (can be many!) and set all the MTUs to match well, this (while maybe on surface helping with the problem, depending on many things) can set one up with a nasty footgun, whereby black hole will show in the most terrible moment of high traffic. See my PMTUD/PLPMTUD rant elsewhere in this thread.
Veserv
MTU discovery would be so much easier if the default behavior was truncate and forward when encountering a oversized packet. The endpoints can then just compare the bytes received against the size encoded inside of the packet to trivially detect truncation and thus get the inbound MTU size.
This allows you to do MTU discovery as a endpoint protocol with all the authentication benefits that provides and allows you to send a single large probe packet to precisely identify the MTU size. It would also allow you to immediately and transparently identify MTU reductions due to route changes or any other such cause instead of packets just randomly blackholing or getting responses from unknown, unauthenticated endpoints.
zamadatix
Truncation for a dedicated probe packet type: you lose the information it's a probe when you go through a tunnel of some sort (VPN, L2TP, IPsec, MPLS, VPLS, VXLAN, PBB, q-in-q, whatever). You're also dealing with different layers e.g. a client could send an L3 packet probe and now you're expecting a layer 2 PBB/q-in-q node to recognize IP packet types and treat them specially (layering violation).
Truncation for all packet types: data in transit can occasionally get split for other reasons. Right now that's just made into loss, if we had built every protocol layer on the idea it should forward anyways then any instances of this type of loss also become MTU renegotiations, at best. At worst we're having to forward generally corrupted packets which can cause all sorts of other problems. It'd be another layering violation to require that e.g. an L2 switch must adjust the UDP checksum when it's intentionally truncating a packet, but that'd be the only way to avoid that. Tunnels (particularly secure) are also tricky here (you need to run multiple separate layers of this continuously to avoid truncation information not propagating to the right endpoints). It also doesn't allow for truly unidirectional protocols e.g. a UDP video stream as there is no allowance for out of session signaling to be possible.
The above is for "if we have started networking day 1 with this plan in mind". There are of course additional problems given we didn't. I'm also not sure I follow how allowing any intermediate node to truncate a packet is any more authenticated.
The (still ugly) beauty of using PMTUD-style approach over truncation or probe+notification is it doesn't try to make assumptions about how anything in the middle could ever work for the rest of time, and that makes it both simple (despite sounding like a lot of work) and reliable. You and your peer just exchange packets until you find the biggest size that fits (or that you care to check for) and you're off! MTU changes due to a path change? No problem, it's just part of your "I had a connection and the other side seems to have stopped responding. How do I attempt to continue" logic (be that retry a new session or attempt to be smart about it). It also plays nice with the ICMP too large messages - if they are there you can choose to listen, if they are not it still "just works".
Or, like the article says, safe minimums can be more practical.
Veserv
You truncate for all packet types.
Data in transit is almost never split for reasons other than fragmentation to avoid MTU problems. Any such split necessarily defines a fragmentation and reconstruction protocol so it still "preserves" the original send length information needed for truncation detection. If they have gone truly crazy and implemented a entire stream protocol transparently backing their flows then their transparent inner point-to-point layer would need to be aware of truncation in much the same way it would need to be aware of MTU limits anyways.
Forwarding generally corrupted packets should not be a problem unless your middleboxes are aggressively engaging in layering violations. From the perspective of a middlebox that is not engaging in layering violations you just have headers with blobs of data. Truncating the blob of data is basically uninteresting; at most you recalculate your integrity tags at your appropriate layer. You do not and should not recompute anything at higher layers. Furthermore, your endpoints must already be robust to blobs of garbage that pass your integrity tag checking because it is trivial for malicious actors to send you blobs of garbage with correctly calculated integrity tags. And, even if you were fully isolated, you can still get correlated bit errors that result in a correct integrity tag despite payload bit errors. Every client implementation that is not grossly incompetent must already be robust to getting garbage. You only get problems when your middleboxes start mucking around and trying to be too smart and violating your point-point transport abstraction.
You still get unidirectional protocols because you should manage truncation information out-of-band of any of your protocols. UDP or any other protocol should not communicate back to the sender that truncation happened. You do that some other way or even do not bother to do it at all. This is extra channel information that you can choose to communicate to let the other endpoint know about channel properties to make better data encoding decisions. You can transmit that in-band, out-of-band, on a different protocol, whatever. This is a higher level property of the communication channel between you and the other side.
Truncation is better authenticated because the packet reaches the other, known, authenticated endpoint who is the entity who can inform you, over a authenticated channel, that the transport channel has problems. You do not get nonsense like ICMP too large messages which come from unknown, unauthenticated entities. Furthermore, truncated messages can still be authenticated as long as you authentication tag the base header which should never be in the truncated section (you still need to have a minimum MTU below which you should always reject, but that number is small and much smaller than existing MTUs).
zamadatix
> Data in transit is almost never split for reasons other than fragmentation to avoid MTU problems
Fragmentation is a specific (unrelated) term, it's not interchangeable with a split. You can have (depending on the protocols involved):
- A runt due to a collision
- A link drop during transmit
- A problem during cut-through type transport
You can do various things to combat some of these (such as fragment-free instead of cut-through in collision domains) but you can't guarantee every phy IP ends up riding over can or should avoid these constraints.
> Forwarding generally corrupted packets should not be a problem unless your middleboxes are aggressively engaging in layering violations. From the perspective of a middlebox that is not engaging in layering violations you just have headers with blobs of data.
If "delivery of something somewhere" is your only definition of a problem, perhaps :p.
> Furthermore, your endpoints must already be robust to blobs of garbage that pass your integrity tag checking because it is trivial for malicious actors to send you blobs of garbage with correctly calculated integrity tags.
Not only the endpoints to garbage in the data payloads but equally the gear to garbage in the network headers. Be it full authentication or just error detection, you don't want to just forward things with a corrupted network header and hope it doesn't cause an issue or security violation. Things like CRCs or HMACs are done per layer precisely for this kind of reason, going to truncation requires dropping that safe handling.
> Every client implementation
As a side note: the concerns have less to do with the clients, they have full context and control of their sessions in software land with little concerns from concerns in being the physical transport layer. Most all of these considerations need to be thought from the intermediate boxes doing the transport/truncation instead.
> You still get unidirectional protocols because you should manage truncation information out-of-band of any of your protocols
Unidirectional protocols cannot be expected to punt directionality to a separate session. In general, any time the answer to a network conundrum (such as the two generals) sounds as easy as "just move that to a separate channel which has the information" you have either duplicated the problem in that channel or added functionality which might not be physically available (or directionally available for security use case reasons, or scalably available for multicast, or something else for a use case that isn't 'inside out' from what might pop in mind as a 'standard' session).
> Truncation is better authenticated because the packet reaches the other, known, authenticated endpoint who is the entity who can inform you, over a authenticated channel, that the transport channel has problems.
I'm still not sure I follow - how is the message between endpoints still authenticated if middleboxes can modify the bytes, breaking an HMAC and/or CRC (if any), and it still gets delivered? Having authenticated an endpoint exists at an address you've sent a packet to before does not automatically authenticate any packet which arrives.
You also skipped over any of the implications for network tunnels (secure/insecure) - is MTU discovery just not supposed to work in those use cases?
I think you can absolutely make a domain specific protocol which is happy to use truncation for MTU discovery, I just don't think anything which is supposed to be as universally usable as IP can.
ay
From the pragmatic standpoint: manually hard coding a safe minimum is the only approach which consistently works.
PMTUD somehow missed that packet networks ditching the OOB mechanisms of circuit switched networks was a good thing. By adding an OOB mechanism of attempted MTU discovery. Unauthenticated.
Yes, matching the 5-tuple from the original payload somewhat helps against the obvious security problem with this. (It was a fun 3-4 years while it was being added to systems across the ‘net while everyone was blocking the ICMP outright to avoid the exploitation. The burps of that one might still find in some security guidelines)
But the number of the network admins who understand what do they have to configure in their ACLs and why, is scarily small compared to the overall pool size.
Here’s another hurdle: for about two decades, to generate ICMP you have to punt the packet from hardware forwarding to the slow path. Which gets rate-limited. Which gives one a fantastic way to create extremely entertaining and hard to debug problems: a single misbehaving or malicious flow can disable the ICMP generation for everyone else.
Make hardware that can do it in fast path ? Even if you don’t punt - you still have to rate-limit to prevent the unauthenticated amplification attack (28 bytes of added headers is not comparable with some of the DNS or NTP scenarios, but not great anyway)
So - practically speaking, it can’t be relied on, other than a source for great stories.
PLPMTUD is a little better, in a sense that it attempts to limit itself to inband probes, but then there is the delicate dance of loss customarily being used to signal the congestion.
So this mechanism isn’t too reliable either, in very painful ways for the poor soul on call dealing with the outcomes. Ask me how I know.. ;-)
Now, let’s add to this the extremely pragmatic and evil hack that is the TCP MSS clamping, coming back from the first PPPoE days; which makes just enough of the eyeball traffic work to make this a “small problem with unimportant traffic that no one cares for anyway”.
So yes, safe minimums are a practical solution.
Until one start to build the tunnels, that is. A wireguard tunnel inside IPSec tunnel. Because policy. Inside VXLAN tunnel inside another IPSec tunnel, because SD-WAN. Which traverses NAT64, because transition and address scarcity.
At which point the previously safe minimums might not be safe anymore and we are back to square 1. I suspect when folks will start running QUIC over wireguard/ipsec/vxlan + IPv6 en masse we will learn that (surprise!) 1200 was not a safe value after all.
So, with this in mind, I posit it’s nice to attempt to at least fantasize about the universe where MTU determination would be done entirely inline, even if hypothetical - if we had the benefit of today’s hindsight and could time travel - could we have made it better ?
P.s. unidirectional protocols could be taken care of by fountain codes not unlike the I-, P- and B- frames in video world, with similar trade offs, moreover, I feel the unequal probability of loss depending on a place in the packet might allow for some interesting tricks.
zamadatix
Agree wholeheartedly on the pragmatic standpoint of just using minimums.
With regard to the problems of out of band signaling in plain PMTUD I fully agree with all your well stated points, doubly so on PLPMTUD! PLPMTUD is my preferred variation of PMTUD and I was glad to see the datagram form utilized in QUIC (especially since it's really a generic secure network tunneling protocol, not just the HTTP variant). I'm also glad QUIC's security model naturally got rid of MSS clamping... it was somewhat pragmatic in one view... but concerning/problematic in others :D. Of course it's not like TCP/mss clamping have exactly gone away though :/.
Also fully agree on both PLPMTUD still not being as reliable/fast as one would like (though I still think it's the best of the options) + safe minimums never seeming to stay "safe". At least IPv6 attempted to hedge this by putting pressure on network admins, saying "everyone is expecting 1280". Of course... we all know that doesn't mean every client ends up with 1280, particularly if they are doing their own VPN tunnel or something, but at least it gives us network guys an extra wall of "well, the standard says we need to allow expectation of 1280 and the rate of bad things which happen will be much higher lower than that".
You seem to have some really neat perspectives on networking, do you mind if I ask about what you do/where you got your experience? I came up through the customer side and eventually over time morphed my way into NOS development at some network OEMs and it feels like I run into fewer and fewer folks who deal with the lower layers of networking as time has went on. I think the most "fun" parts are trying to design overlay/tunneling systems which are hardware compatible with existing ASICs or protocols but are able to squeeze some more cleverness out of the usage (or, as you put it, if we had the benefit of today’s hindsight and could time travel - could we have made it better). The area I'd say I've been least involved in, but would like to, is anything to do with time sensitive networking or lossless ethernet use cases.
ay
With IPv4, clearing the DF bit in all egress packets and hacking on top of QUIC could give just enough of a wiggle room to make it possible to explore this between a pair of cooperating hosts even in today’s Internet.
Anti-DDoS middle boxes will be almost certainly unhappy with lone fragments and UDP in general, so it’s a bit of a thorny path.
The big question is what to do with IPv6, since the intermediary nodes will only drop. This bit unfortunately makes the whole exercise pretty theoretical, but it can be fun nonetheless to explore.
Feel free to contact me at my github userid at gmail, if this is a topic of interest.
zamadatix
Most carrier/enterprise/hardware IPv4 routers, particular those on the internet, will not actually perform IPv4 fragmentation on behalf of the client traffic even though it's allowed by the IPv4 standard. Typically fragmentation is reserved for boxes which already have another reason to care about it (such as needing to NAT or inspect the packets) or the client endpoints themselves. I.e. the internet will (sparing security middleboxes) allow arbitrary IPv4 fragments through but it won't typically turn a 8000 byte packet into 6 fragments to fit through a 1500 byte MTU limitation on behalf of the clients. E.g. if you send a 1500 byte IPv4 ping without DF set to a cellular modem or someone with a DSL modem using PPPoE it'll almost always get dropped by the carrier rather than fragmented.
Of course nothing is stopping you from labbing it up at home. Firewalls and software routers can usually be made to do refragmentation.
ay
Of course on the carrier boxes the fragmentation is done also not inline, so its behavior will depend on the aggressiveness of the CoPP configuration, and will be subject to the same pitfalls as the ICMP packet too big generation.
Thanks for keeping me straight here!
Based on the admittedly old study at [0] seems like some carriers just don’t bother to fragment, indeed - but by far not all of them.
Firewalls might do virtual reassembly, so the trick with the initial fragment won’t fly there.
This MTU subject is interesting for me because I have a little work in progress experiment: https://gerrit.fd.io/r/c/vpp/+/41914/1/src/plugins/pvti/pvti... (the code itself is already in, but has a few crashy bugs still and I need to take make it not suck performance wise, but that is my attempt to revisit the issue of MTU for tunnel use case. The thesis is that keeping the 5-tuple will make “chunking”/“de-chunking” at tunnel endpoints much much simpler on the endpoints of the tunnel.
The source of inspiration was a very practical setup at [1], which is, while looking horrible in theory (locally fragmented GRE over L2TP), actually gives a decent performance with 1500-byte end to end MTU over the tunnel.
The open question is which inner MTU will be sane, taking into account the increased probability of loss with bigger inner MTU… intuitively seems like something like ~2.5K should just double the loss probability (because it’s 2x packets) and might be a workable compromise in 2025….
One could also do the same trick over QUIC, of course, but i wanted something tiny and easier to experiment with - and the ability to go over IPSec or wireguard as well as a secured underlay.
[0] https://labs.ripe.net/author/emileaben/ripe-atlas-packet-siz...
ikiris
And how do you tell the difference between cut off packets, and a mtu drop? What about crcs / frame checks? Do you regenerate the frames? Do you do this at routed interfaces? What if there's just layer 2 only involved?
LegionMammal978
> And how do you tell the difference between cut off packets, and a mtu drop?
You don't, apart from enforcing a bare-minimum MTU for sanity's sake. If your jumbo-size packets are getting randomly cut off by a middlebox, then they probably aren't stable at that size anyway.
Veserv
Packets do not get “cut-off” normally. That is kind of the point. Some protocols allow transparent fragmentation, but the fragments need to encode enough information for reconstruction, so you can still detect “less data received than encoded on send”.
You do not need bit error detection because you literally truncated the packet. The data is already lost. But in the process you learned it was due to MTU limits which is very useful. Protocols are already required to be robust to garbage that fails bit error detection anyways, so it is not “required” to always have valid integrity tags. You could transparently re-encode bit error detection on the truncated packet if you so desire to ensure data integrity of the “MTU resulted in truncation” packet that you are now forwarding, but again, not necessary.
Any end-to-end protocol that encodes the intended data size in-band can use this technique across truncating transport layers. And any protocol which does so already requires implementations to not blindly trust the in-band value otherwise you get trivial buffer overflows. So, all non-grossly insecure client implementations should already be able to safely handle MTU truncation if they received it (they would just not be able to use that for MTU discovery until they are updated). The only thing you need is routers to truncate instead of drop and then you can slowly update client implementations to take advantage of the new feature since this middlebox change should not break any existing implementations unless they are inexcusably insecure.
ikiris
I don’t think you understand what normally looks like if you start forwarding damaged frames like this because you can’t tell the difference. That was the point.
cryptonector
> Path MTU discovery has not been enthusiastically embraced
Ugh. I don't understand this. Especially passive PMTUD should just be rolled out everywhere. On Linux it still defaults to disabled! https://sourcegraph.com/search?q=context%3Aglobal+repo%3A%5E...
whiatp
PMTU just doesn't feel reliable to me because of poorly behaved boxes in the middle. The worst offender I've had to deal with was AWS Transit Gateway, which just doesn't bother sending ICMP too big messages. The second worst offender is, IMO (data center and ISP) routers that generate ICMP replies in their CPU, meaning large packets hit a rate limited exception punt path out of the switch ASIC over to the cheapest CPU they could find to put in the box. If too many people are hitting that path at the same time, (maybe) no reply for you.
More rare cases, but really frustrating to debug was when we had an L2 switch in the path with lower MTU than the routers it was joining together. Without an IP level stack, there is no generation of ICMP messages and that thing just ate larger packets. The even stranger case was when there was a Linux box doing forwarding that had segment offload left on. It was taking in several 1500 byte TCP packets from one side, smashing them into ~9000 byte monsters, and then tried to send those over a VPNish network interface that absolutely couldn't handle that. Even if the network in the middle bothered to generate the ICMP too big message, the source would have been thoroughly confused because it never sent anything over 1500.
toast0
> The even stranger case was when there was a Linux box doing forwarding that had segment offload left on. It was taking in several 1500 byte TCP packets from one side, smashing them into ~9000 byte monsters, and then tried to send those over a VPNish network interface that absolutely couldn't handle that. Even if the network in the middle bothered to generate the ICMP too big message, the source would have been thoroughly confused because it never sent anything over 1500.
This is an old Linux tcp offloading bug; large receive offload smooshes the inbound packet, then it's too big to forward.
I had to track down the other side of this. FreeBSD used to resend the whole send queue if it got a too big message, even if the size did not change. Sending all at once made it pretty likely for the broken forwarder to get packets close enough to do LRO, which resulted in large enough packet sending to show up as network problems.
I don't remember where the forwarder seemed to be, somewhere far away, IIRC.
cryptonector
> PMTU just doesn't feel reliable to me because of poorly behaved boxes in the middle. The worst offender I've had to deal with was AWS Transit Gateway, which just doesn't bother sending ICMP too big messages.
Passive PMTUD does NOT depend on ICMP messages.
Hikikomori
They recently started supporting pmtud on tgw. But it wasn't a big deal really as it adjusted mss instead.
immibis
L2 not generating errors is expected behaviour - all ports on the L2 network are supposed to have the same MTU set
mkj
Would that help with UDP, or only TCP?
ajb
That particular one, only TCP. There is a different one for UDP applications: https://www.rfc-editor.org/rfc/rfc8899
Because UDP is only a very thin layer, each layer on top (eg, QUIC) has to implement PLPMTUD; although, recently IETF standardised a way to extend UDP to have options and PLPTMUD is also specified for that: https://datatracker.ietf.org/doc/draft-ietf-tsvwg-udp-option...
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cryptonector
You can implement passive PMTUD with UDP if you like. It's more work for you, but it's perfectly doable.
posnet
"Jumbogram", an IPv6 packet with the Jumbo Payload option set, allowing for an frame size of up to 2³²-1 bytes.
At 10Gbps it would take 3.4 seconds just to serialize the frame.
hugmynutus
Luckily 400Gb/s nics are already on the market [1]
roeles
> The system dispensed with a passive common bus and replaced it with an active switching hub to which hosts were attached.
I get the impression that the standard still allows hubs to exist, but that you just don't see them in practice.
I would be interested if anyone has ever used a 100mbit hub.
jiehong
No commitee want to change it, because nobody agrees. And nothing changes.
Can’t we accept to start a change that may take a decade or more to go forward? Instead of not starting that change.
nayuki
> The speed of light in glass or fiber-optic cable is significantly slower, at approximately 194,865 kilometers per second. The speed of voltage propagation in copper is 224,844 kilometres per second.
If I understand correctly, the speed of light in an electrical cable doesn't depend on the metal that carries current, but instead depends on the dielectric materials (plastic, air, etc.) between the two conductors?
tonyarkles
If I’m interpreting what you’re asking correctly, yes. The velocity factor of a cable doesn’t spend on the metal it’s made of but rather the insulator material and the geometry of the cable.
For fibre the velocity factor depends on the refraction index of the fibre.
lucb1e
Huh? Maybe I'm completely misreading the question, but when they say fiber-optic cable, they do mean optic. It's not an "electrical cable"; there is no metal needed in optic communication cables (perhaps for stiffness or whatnot, but not for the communication)
Hikikomori
>The speed of voltage propagation in copper is 224,844 kilometres per second.
This part?
lucb1e
What about it?
beeburrt
That font size is tiny. If this is your site, maybe consider a larger font size
nayuki
The site specifies a base font size of 12px. The better practice is to not specify a base font size at all, just taking it from the user's web browser instead. Then, the web designer should specify every other font size and box dimension as a scaled version of the base font size, using units like em/rem/%, not px.
Related reading: https://joshcollinsworth.com/blog/never-use-px-for-font-size
lucb1e
It's the same size as HN: 12px. HN looks larger to me for some reason, but I can't figure out why: when I overlay a quote someone posted here over the website with half transparency in GIMP, the text is clearly the same height. Some letters are wider, some narrower, but the final length of the 8 words I sampled is 360px on HN vs. 358px on that website (so differences basically cancel out)
This is on Firefox/Debian, in case that means something for installed fonts. I see that site's CSS specifies Verdana and Arial, names that sound windowsey to me but I have no idea if my system has (analogous versions to) those
tomthecreator
There's a PDF version linked at the top of the article, it's actually much better typeset.
usefulcat
Given the subject of TFA, this seems appropriate in a meta sort of way.
nullc
Is there any convenient way to tell linux distributions that the local subnet can handle 9k jumbos (or whatever) but that anything routed out must be 1500?
I currently have this solved by just sticking hosts on two vlans, one that has the default route and another that only has the jumbo capable hosts. ... but this seems kinda stupid.
fbouynot
Yes you can set your interface MTU at 9000 and assign a 1500 MTU to the routes themselves.
throw0101b
> […] and assign a 1500 MTU to the routes themselves.
See "mtu" option in ip-route(8):
* https://man.archlinux.org/man/ip-route.8.en#mtu
The BSDs also have an "-mtu" option in route(8):
2OEH8eoCRo0
Do you count the frame preamble?
jeffbee
The efficiency argument applies to private flows mostly. In terms of overall network traffic, the huge majority takes place between peers that share a local or private network. Internetworking as such has a relatively small share of total flows. So large frame sizes are beneficial in the context where they are also not problematic, and path MTU discovery is not beneficial in the context where it has many drawbacks. It seems as though the current state is pretty much optimal.
yb303
tldr- a document written in 2024 that does fit on my phone
This reminds me of one of the most interesting bugs I've faced: I was responsible for developing the component that provided away market data to the core trading system of a major US exchange (which allows the trading system to determine whether an order should be matched in-house or routed to another exchange with a better price).
Throughputs were in the multiple tens of thousands of transactions per second and latencies were in single digit milliseconds (in later years these would drop to double digit microseconds, but that's a different story). Components were written in C++, running on Linux. The machine that ran my component and the trading engine were neighbors in a LAN.
We put my component through a full battery of performance tests, and for a while, we seem to be meeting the numbers. Then one day, with absolutely zero code changes from my end or the trading engine's end, the the latency numbers collapsed. We checked the hardware configs and the rate at which the latest test was run. Both identical.
It took, I think, several days to solve the mystery: in the latest test run, we had added one extra away market to a list of 7 or 8 markets for which my component provided market data to the trading system. We had added markets before without an issue. It's a negligible change to the market data message size, because it only adds a few bytes: market ID, best bid price & quantity, best offer price & quantity. In no way should such a small change result in a disproportionate collapse in the latency numbers. It took a while for us to realize that before the addition of these few bytes, our market data message (a binary packed format), neatly fit into a single ethernet frame. Those extra few bytes pushed it over the 1600 (or 1500?) mark and caused all market data message frames (which were the bulk of messages on the system, next to orders), to fragment. The frame fragmentation and reassembly overhead was enough to clog up the pipes at the rates we were pumping data.
In the short run, I think we managed to do some tweaks and get the message back under 1600 bytes (by omitting markets that did not have a current bid/offer, rather than sending NULLs). I can't recall what we did in the long run.