Hi Carsten, thanks for your insights.
> On Mar 17, 2019, at 15:34, Carsten Bormann <[email protected]> wrote: > >>> The end-to-end argument applies: Ultimately, there needs to be >>> resequencing at the end anyway, so any reordering in the network would be a >>> performance optimization. It turns out that keeping packets lying around >>> in some buffer somewhere in the network just to do resequencing before they >>> exit an L2 domain (or a tunnel) is a pessimization, not an optimization. >> >> I do not buy the end to end argument here, because in the extreme why >> do ARQ on individual links anyway, we can just leave it to the end-points to >> do the ARQ and TCP does anyway. > > The optimization is that the retransmission on a single link (or within a > path segment, which is what I’m interested in) does not need to span the > entire end-to-end path. That is strictly better than an end-to-end > retransmission. I agree, and by the same logic local resequencing is also better, unless the re-ordering event happened at the bottleneck link. > Also, a local segment may allow faster recovery by not implicating the entire > e2e latency, which allows for strictly better latency. > So, yes, there are significant optimizations in doing local retransmissions, > but there are also interesting interactions with end-to-end retransmission > that need to be taken care of. This has been known for a long time, e.g., > see https://tools.ietf.org/html/rfc3819#section-8 which documents things that > were considered to be well known in the early 2000s. Thanks, but my understanding of this is basically that a link should just drop a packet unless it can be retransmitted with reasonable effort (like the G.INP retransmissiond on dsl-links will give up); sure we can argue about what "reasonable effort" is in reality, but I fear if we move away from 3 dupACKs to say X ms all transport links will assume they have leewway to allow re-ordering close to X, that will certainly be worse than today. And since I am an end-user and do not operate a transport network, I know what I prefer here... > >> The point is transport-ARQ allows to use link technologies that otherwise >> would not be acceptable at all. So doing ARQ on the individual links already >> indicates that somethings are more efficient to not only do e2e. > > Obviously. > >> I just happen to think that re-ordering falls into the same category, at >> least for users stuck behind a slow link as is typical at the edge of the >> internet. > > Resequencing (which is the term I prefer for putting things back in sequence > again, after they have been reordered) requires storing packets that are > ahead of later packets. Obviously. > This is strictly suboptimal if these packets could be delivered instead (in > contrast, it *is* a good idea to resequence packets that are in a queue > waiting for a transmission opportunity). Fair enough, but that basically expects the bottleneck link that actually accumulates a queue to do the heavy lifting, not sure that the economic incentives are properly aligned here. > So *requiring*(*) local path segments to resequence is strictly suboptimal. > > (*) even if this is not a strict requirement, but just a statement of the > form “the transport will be much more efficient if you deliver in order”. My point is the transport will much more useful if if undertakes (reasonable) effort to deliver in-order, that is slight;y different, and I understand that those responsible for transport networks have a different viewpoint on this. > >> To put numbers to my example, assume I am on a 1/1 Mbps link and I get TCP >> data at 1 Mbps rate and MTU1500 packets (I am going to keep the numbers >> approximate) and I get a burst of say 10 packets containing say 10 >> individual messages for my application telling the position of say an object >> in 3d space >> >> each packet is going to "hog" the link for: 1000 ms/s * (1500 * 8 b/packet ) >> / (1000 * 1000 b/s) = 12 ms >> So I get access to messages/new positions every 12 ms and I can display this >> smoothly > > That is already broken by design. Does not matter much, a well designed network should also allow to do stupid things... > If you are not accounting for latency variation (“jitter”), you won’t be > able to deal with it. Which would just complicate the issue a bit if we would introduce a say 25 ms de-jitter buffer without affecting the gist of it. > Your example also makes sure it does not work well by being based on 100 % > utilization. Same here, access links certainly run closer to 100% utilization than core links, so operation at full saturation is not completely unrealistic, but I really just set it up that way for clarity. > >> Now if the first packet gets r-odered to be last, I either drop that packet > > …which is another nice function the network could do for you before expending > further resources on useless delivery; see e.g. draft-ietf-6lo-deadline-time > for one way to do this. Yes, but typically I do not want the network to do this, as I would be quite interested in knowing how much too late the packet arrived. > >> and accept a 12 ms gap or if that is not an option I get to wait 9*12 = >> 108ms before positions can be updated, that IMHO shows why re-ordering is >> terrible even if TCP would be more tolerant. > > You are assuming that the network can magically resequence a packet into > place that it does not have. All I expect is that the network makes a reasonable effort to undo re-ordering close to where re-ordering happened. > > Now I do understand that forwarding an out-of-order packet will block the > output port for the time needed to serialize it. So if you get it right > before what would have been an in-order packet, the latter incurs additional > latency. Note that this requires a bottleneck configuration, i.e., packets > to be forwarded arrive faster than they can be serialized out. Don’t do > bottlenecks if you want ultra-low latency. (And don’t do links where you > need to retransmit, either.) I agree, but that is live with a home internet access link, the bottleneck is there. This also points out a problem with the L4S argument for end-users, as the ultra-low latency (their words, not mine) will not realize for end-users close to what the project seems to promise. > >> Especially in the context of L4S something like this seems to be totally >> unacceptable if ultra-low latency is supposed to be anything more than >> marketing. > > Dropping packets that can’t be used anyway is strictly better than delivering > them. Well, not for L4S, as TCP Praque is supposed to fall back to legacy congestion control behavior upon encountering packet drops... > But apart from that, forwarding packets that I have is strictly better for > low latency than leaving the output port idle and waiting for > previous-in-order packets to send them out in sequence. It really depends what we mean when we talk about latency here, as shown for and end-user that might be quite different... > >>> For three decades now, we have acted as if there is no cost for in-order >>> delivery from L2 — not because that is true, but because deployed transport >>> protocol implementations were built and tested with simple links that don’t >>> reorder. >> >> Well, that is similar to the argument for performing non-aligned loads >> fast in hardware, yes this comes with a considerable cost in complexity and >> it is harder to make this go fast than just allowing aligned loads and >> fixing up unaligned loads by trapping to software, but from a user >> perspective the fast hardware beats the fickle only make aligned loads go >> fast approach any old day. > > CPUs have an abundance of transistors you can throw at this problem so the > support of unaligned loads has become standard practice for CPUs with enough > transistors. > I’m not sure this argument transfers, because this is not about transistors > (except maybe when we talk about in-queue resequencing, which would be a nice > feature if we had information in the packets to allow it). Like the 5-tuple in TCP and UDP? This example was not meant to taken literally, but just to illustrate that depending on the level of observation speeding up one domain can have an noticeable effect on another one, but might still be worth the effort. > >>> Techniques for ECMP (equal-cost multi-path) have been developed that >>> appease that illusion, but they actually also are pessimizations at least >>> in some cases. >> >> Sure, but if I understand correctly, this is partly due to the fact >> that transport people opted not to do the re-sorting on a flow-by-flow >> basis; that would solve the blocking issue from the transport perspective, >> sure the affected flow would still suffer from some increased delay, but as >> I tried to show above that might be still smaller than the delay incurred by >> doing the re-sorting after the bottleneck link. What is wrong with my >> analysis? > > Transport people have no control over what is happening in the network, so > maybe I don’t understand the argument. If the remote end of a potentially re-ordering link would implement fair-queueing (a big if, sure) then it should be easy to only stall the flows that have outstanding packets, and this could be solely be based on the local retransmit ACKs so the link would only need to clean up its own re-orderings an could just faithfully relay a flow that entered the link already with re-ordering. This might in reality not be feasible at all... > >>> The question at hand is whether we can make the move back to end-to-end >>> resequencing techniques that work well, >> >> But we can not, we can make TCP more robust, but what I predict if RACK >> allows for 100ms delay transports will take this as the new the new goal and >> will keep pushing against that limit; and all in the name of bandwidth over >> latency. > > Where does this number come from? 100 ms is pretty long as a reordering > maximum for most paths outside of satellite links. Instead, you would do > something based on an RTT estimate. I just made that number up as the exact N does not matter, the argument is what ever we set as the new threshold will be approached by transport characteristics. Then again havin something that inversely scales with bandwidth is certainly terrible from a transport perspective, so I can understand the argument for a fixed temporal threshold. > >>> at least within some limits that we still have to find. >>> That probably requires some evolution at the end-to-end transport >>> implementation layer. We are in a better position to make that happen than >>> we have been for a long time. >> >> Probably true, but also not very attractive from an end-user >> perspective…. unless this will allow transport innovations that will allow >> massively more bandwidth at a smallish latency cost. > > The argument against in-network resequencing is mostly a latency argument > (but, as a second order effect, that reduced latency may also allow more > throughput), so, again, I don’t quite understand. As I tried to show for TCP the flow with re-ordered packets certainly pays a latency cost that especially if re-ordering does not happen on the bottleneck link but at a faster link could be smaller. Gruss Sebastian > > Grüße, Carsten > _______________________________________________ Bloat mailing list [email protected] https://lists.bufferbloat.net/listinfo/bloat
