Comment by fn-mote | Hacker News Reader

Comment by fn-mote | Hacker News Reader

This article was interesting and also frustrating to read.

1. There are very few numbers. In particular, improvement in performance under various circumstances is _not_ given! If you dig around you can find their transfer time application [1], but there is no discussion on that page.

2. The basis for the improvement is not spelled out. (References are given, but you have to know the field - "acronyms only".) If I understand correctly, their contribution is the improved measures of congestion used. Their landing page just touts "don't use TCP"... which sounds like Step 0 of a very long process.

I admit, the title is basically accurate: "how to build" not "the performance of".

tl;dr: Start with existing work, simulate and improve incrementally.

I don't know anything about the field, but this article didn't lead me to understand any better. I'd love to know the real numbers they observed, which approaches didn't pan out, are they effectively using an error correcting code?

Anyway, it's certainly not an academic paper - just an advertisement.

[1] https://www.trytachyon.com/file-transfer-calculator

I just tried the calculator. It seems that if you're in "US Metro" or "Europe", then the transfer protocol is just as fast as TCP, is this correct? I wonder why this is the case. Is it because the routers play more fairly?

I would expect it means your service provider isn't dropping packets. Their protocol seems to just be more aggressive about not backing off in the face of packet loss, which is helpful if one of your links is a marginal radio connection.

The cynic in me thinks they achieve better throughput because they don't play nice with TCP and monopolize the link while everybody else gets backed off.

I'm inclined to agree. Perhaps there's a place for a "fast" FTP protocol but I wouldn't want to be sharing the link with them ... imagine the cries going around the office ... "has the internet gone down again!?"

For TCP transfers, I expect to see near 100% utilisation (if I'm not sharing bandwidth) but that's on wired (reliable) links.

nailed it.

communication links are generally a shared resource (common good) and the experience will degrade for everyone if individual transfers/users aren't backing down voluntarily.

You can do better than TCP and still "play nice". Just using up exactly the bandwidth TCP refuses to use, as FASP does by default, gets you much faster transmission.

> Just using up exactly the bandwidth TCP refuses to use

There's not much good to say about ppl that "use up the space others refuse to use" at the queue in a store, thou.

i.e. our network links are generally not designed to be used at full capacity b/c queueing is hard and buying faster/more gear is easy.

Then let's not say it. Clearly you miss the point: any provisioned backbone bandwidth that does not carry a packet just goes to waste. Network backbone channels are not provisioned like your consumer-grade cable TV network endpoint hubs. Carriers pay for guaranteed capacity. When TCP fails to use bandwidth capacity, it is failing to use backbone bandwidth, not dodgy last-mile bandwidth.

Even moreso: when so configured, FASP is having exactly zero effect on the actual bit rate of the TCP connections. Another TCP connection would compete for the same packet slots, slowing them all down. The FASP traffic does not slow down any of the TCP connections, but only competes with other FASP (or similar) traffic.

Anything that causes packet loss will slow down TCP. If you are fully saturating a link you will introduce packet loss. It's just how TCP works.

And if you are fully saturating exactly the packet slots TCP would not try to use, then you are not introducing packet loss. That is just how FASP works.

there is just no way for end-points to know about the 'unused slots' much less use them specifically (w/o introducing packet loss or latency).

> The cynic in me thinks they achieve better throughput because they don't play nice with TCP and monopolize the link while everybody else gets backed off.

sounds like BBRv1

Thanks for taking the time to read it! To address your concerns:

1. To give you an idea of the speed improvements, we transferred a 2GB file between Ohio and Singapore on AWS and were able to transfer it in 0:26 (seconds) using our protocol, vs 2:15 for SCP.

2. The basis for improvement is taking into account the changes in round-trip-time for a particular network path; these temporary increases are used as the primary congestion signal.

We are not using error correcting codes, which are good for preventing the retransmission of packets but do not address the underlying problem of avoiding congestion in a network.

How much was SCP affected by TCP buffer size tuning?

Can I ask a dumb question? Why SCP and not rsync?

There is no meaningful difference, at the end of the day they'll both be fine for measuring the base case of single-threaded TCP connection performance.

I prefer rsync as well, but SCP works on more machines by default (no dependency on rsync being installed on the target host).

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