Can you elaborate more on why applications handle packet reordering badly? Is it just that applications are written to assume in-order packets are the "fast path" and need to use less optimized code when things come in out-of-order, or is there some other heuristics that assume packet reordering is rare that would misfire here?

TCP is going to be not great because when you receive packets out of order, there's a feedback loop that leads to the source retransmitting the presumed missing, but possibly just delayed packets. Adding more duplicate data to your congested queue makes things worse / keeps them bad. It's much better to drop some packets from the flow than to intentionally misorder them. This property does make it harder to load balance tcp over multiple connections; you get better results if you hash flows, so one tcp connection always goes over the same physical connection, and should remain ordered... Round robin packet sending might be nice, since you wouldn't end up with uneven utilization of individual physical connections in an aggregated connection, but the tcp stack behavior would be much worse and real world bandwidth would not be useful. Actually handling tcp packets out of order isn't nice either, but if the options are drop a packet or delay an earlier packet to send after a later one, we're going to have to handle some packets out of order. This is what ECN (explicit congestion notification) was supposed to help, if you could mark packets as congested before the congestion got bad enough to drop packets; of course, that was hard to get deployed.

Things like voip or gaming are going to have trouble with out of order packets too. If you already did something to workaround the missing packet, if it eventually arrives late, you may not have any use for it. If I already played silence (or ??) to get through a missed sample, I can't go back and put in the late sample. Etc. Late at that point is not better than never.

There are certainly ways to make protocols where late is better than never; you could have a bulk file transfer protocol that sent all data once before resending or something, but that's not common.

I think TCP's logic is more subtle than "if anything comes in out of order, retransmit", in particular, there's a "Retransmission Timeout" that converges to something like ~2x the average roundtrip latency. I assume that custom UDP-based protocols used by VoIP and such will have similar provisions, where they don't declare a packet "lost" until a significant period of time has elapsed. My understanding is that the reason these mechanisms don't handle multiple connections well is because the two connections might have dramatically different typical latencies, not because the packets may be mildly misordered on the other end.

I'm not sure if more modern "smart" protocols that themselves explicitly try to measure and model the underlying network buffers would be confused by LIFO though, especially since the variance in round trip latency will be higher with a LIFO queue.

https://www.catchpoint.com/blog/tcp-rtt

Looking at voip specifically, you've got what's called a jitter buffer, which lets you collect late packets, either out of order or delayed, and still use them, but delay is bad for user experience, so you tend to make that buffer as small as you can. You don't retransmit voip packets, because if they're not fresh, they're not useful. LIFO doesn't make sense for voip, because either you run a very long jitter buffer and nobody wants to talk with you like that, or the delayed packets can't be used.

The tcp retransmit timer is used when you send a packet (or packets) and don't get any acknowledgements. But if you send many packets, and the peer misses one (because it's delayed/out of order), it will send an ack of the last in order sequence (and hopefully selective ack too, it's 2022). If you recieve enough acks of the same packet, that triggers fast retransmit; by rfc2001 and updates, three duplicate acks is the threshold to retransmit, without waiting for the retransmit timer. LIFO would significantly harm the network in case of bursty traffic: if my flow sends 5 packets back to back (which is common with tcp segmentation offloading), and they get queued, they'll get sent to the peer in the exact wrong order, and thaf peer will send the same ack for the first foud packets, before sending a new ack on the fifth. My side will get those first four acks, and retransmit the first packet of the burst, then get the fifth ack, and maybe release a new burst. That's an extra full packet, plus it's typical to only send every other ack normally, so that's extra acks on the return side. Dropping a packet in the flow would still result in extra acks though, but the retransmitted data packet wouldn't be a duplicate through that bottleneck, because the first one was dropped before the bottleneck.

Makes sense, thanks for the info.