Slashdot Mirror
Replacing TCP?
olau writes "TCP, the transfer protocol that most of the Internet is using, is getting old. These guys have invented an alternative that combines UDP with rateless erasure codes, which means that packets do not have to be resent. Cool stuff! It also has applications for peer-to-peer networks (e.g. for something like BitTorrent). They are even preparing RFCs! The guy who started it, Petar Maymounkov, is of Kademlia fame."
Now that Digital is little more than IP spread across a few different companies, maybe the holder of Decnet's patents could release the protocol under an Open Source license. If I recall correctly it was quite the networking layer.
BLING BLING. Meet the architecture that's changing everything.
TCP may be old, but it can go on for another 50 years wothout any problem.
--- "To pee or not to pee, that is the question." ---
Does it have Evil Bit implemented?
http://en.wikipedia.org/wiki/Kademlia
The submitter says that TCP is getting old, but does that really tell us anything about how well it does its job?
If it's not an open protocol (if they charge for use) it may find niche applications. If it is, it may proliferate. I wasn't able to find details about this on the site.
The World Wide Web is dying. Soon, we shall have only the Internet.
Some inefficiencies are one thing, but you're going to need a compelling reason to get everyone to switch.
TCP is old, but that doesn't mean it's bad or replacement is due. Some shortcomings have surfaced and been adressed. For the most part, TCP does a good job at what it was designed to do.
Please correct me if I got my facts wrong.
Because then you're going to have the suits trying to push it down, no matter how great/useful it is in an effort to kill the possibility of coming out with something that could make pirating any easier or more efficient. That's the only way they're going to see it.
It's good to see innovation though, nonetheless.
This is hardly an innovative idea, and usually by the time you end up considering all the issues you wind up with something that looks a lot more like TCP than you had originally intended.
Plus there are already protocol stacks that work around most of their gripes about TCP (slow performance over long pipes, etc).
Here is a summary of their technology copied from their website:
Rateless Internet The Problem
Rateless Internet is an Internet transport protocol implemented over UDP, meant as a better replacement of TCP. TCP's legacy design carries a number of inefficiencies, the most prominent of which is its inability to utilize most modern links' bandwidth. This problem stems from the fact that TCP calculates the congestion of the channel based on its round-trip time. The round-trim time, however, reflects not only the congestion level, but also the physical length of the connection. This is precisely why TCP is inherently unable to reach optimal speeds on long high-bandwidth connections.
A secondary, but just as impairing, property of TCP is its inability to tolerate even small amounts (1% - 3%) of packet loss. This additionally forces TCP to work at safe and relatively low transmission speeds with under 1% loss rates. Nevertheless, our extended real-life measurements show that highest throughput is generally achieved at speeds with anywhere between 3% and 5% loss.
The Solution
By using our core coding technology we were able to design a reliable Internet transmission protocol which can circumvent both of the fore-mentioned deficiencies of TCP, while still remaining TCP-friendly. By using encoded, rather than plain, transmission we are able to send at speeds with any packet loss level. Rateless coding is used in conjunction with our Universal Congestion Control algorithm, which allows Rateless Internet to remain friendly to TCP and other congestion-aware protocols.
Universal Congestion Control is an algorithm for transmission speed control. It is based on a simple and clean idea. Speed is varied in a wave-like fashion. The top of the wave achieves near-optimal throughput, while the bottom is low enough to let coexisting protocols like TCP smoothly receive a fair share of bandwidth. The time lengths of the peaks and troughs can be adjusted parametrically to achieve customized levels of fairness between Rateless Internet and TCP.
The Rateless Internet transport is now available through our Rateless Socket product in the form of a C/C++ socket library. Rateless Internet is ideal for Internet-based applications, running on the network edges, that require high bandwidth in a heterogenous environment. It was specifically built with peer-to-peer and live multimedia content delivery applications in mind.
A slightly faster equivelent to TCP that I have to pay for and no-one else uses.
Sign me up for that sucker right now.
http://jfin.org/jFin pure java open source financial library
Considering this is slashdot and all, I was surprised that their implementation does not appear to be open source (or indeed, freely available at all), though presumably such an implementation will be possible following the RFCs. It seems to work nicely alongside TCP using UDP, quite a cool idea. The question is whether it can break TCP's de facto stranglehold on reliable Internet communication. I'd love to play with it if I could.
Often stories are posted that refer to products or code names, with no description, which is quite annoying.
I'm glad to see this post doesn't run that risk.
Thanks for clearing that up for me.
-d
Love many, trust a few, do harm to none.
There's no doubt that an alternative to TCP might have technical merits. But as far as communication protocols go, TCP itself is pretty amazing. Modern TCP implementations have been tweaked over decades and have impressive performance and reliability. And modern TCP/IP stacks have rather unspoofable connection establishment, another excellent feature for security.
If you want to replace TCP, you have to do more than just develop a new protocol that is faster. It would have to outperform TCP in speed, reliability, and substantially so in order to outweigh the costs of ditching a well-established and trusted protocol.
While this sounds very interesting (have to re-take all those networking certification exams again, I guess), when I read this...
...my eyes told my brain this...
The guy who started it, Petar Maymounkov, is of Kademlia fame."
The guy who started it, Petar Maymounkov, is of Chlamydia fame."
I was about to wonder what sort of "fame" you could get from that. Need coffee. Need sleep.
IronChefMorimoto
R-E-S-P-E-C-T
Find out what it means to me
R-E-S-P-E-C-T
Take care, TCP
Oh socket to me, socket to me,
socket to me, socket to me...
Do really dense people warp space more than others?
Can anybody explain to me how this technology? It reads like marketing speak to me, despite the fact that it will be/is released as open source. How does the technology actually achieve reliability without retransmits? Does it actually achieve it?
Please correct me if I got my facts wrong.
1. This is coming from a company who are surely going to want to make money out of it somehow. Part of the reason TCP succeeded is there was no one to pay.
2. They don't seem to understand the GPL:
"We are planning to release Rateless Codes for free, non-commercial use, in open source under the GNU Public License."
The GPL doesn't restrict commercial use, and hence the only way that they can do this is either they try to add some conditions to the GPL, or they use another mechanism to restrict commercial use: e.g. patents.
No matter how good this technology is it's not going to get wide adoption is an alternative to TCP unless it's unencumbered.
John.
So you say if telcos sell more bandwith than their infrastructure can handle there will be problems...
Uh. Wouldnt have thought THAT could happen....
HI O WISE PRINCE. WHT TOOK U SO DAM LONG?
I did read their website and it looks like their revolutionary new replacement for TCP is UDP with their proprietary ECC built on top of it. However, there is a good reason why TCP never used ECC (they did exist back then).
1) The major problem a TCP packet will face is getting dropped. They mention this problem. They claim their encoding will solve this problem. It won't. No ECC algorithm will allow you to recover a dropped packet.
2) Most packets that are corrupted are corrupted well beyond the repair of most ECCs.
3) ECCs will cause packet size to increase. Not a huge problem, but why do it when ECCs don't help too much to begin with?
I fail to see what is flamebaiting it is to say that TCP can go on for another 50 years, without problem.
Exactly the same kind of post a bit below gets 'insightful'.
It is simply true. Yes, there are some little drawbacks with TCP, but in the whole article, they do not give a compelling reason to switch, let alone why one would *have* to. I mean, RTFA: TCP is at 1-3% and the most efficient would be a throughput with 3-5% (loss)...but so what? It's not optimal, but does it anywhere claims TCP is doomed because it's not optimal in certain area's?
There are myriads of things that aren't optimal on the Net, it doesn't mean they have been here for years and will be for years to come, nor that it is a necessity to switch, if the only thing lacking is that it's not optimally suited.
--- "To pee or not to pee, that is the question." ---
the real issue with any new technology like this is will it be accepted as a standard? For that to happen it needs to be relyable, easy to get (free), and hard to exploit. This looks to have all the above so in 5 to 10 years maybe we'll see this implmented everywhere.
It doesn't seem reason enough to build a new protocol for TCP replacement just because it is "old". Protols ar not living beings you know, aging doesn't show on their cells. Of course troubles always has been popping up over the years, but nothing unsolvable (and nothing in the likings of IP).
:D ) is a bit winged to say the least.
All in all, it's good to have new alternative solutions and new technologies at hand, but to state such things as replacing TCP 'cause its age (maturity, that is
I am putting myself to the fullest possible use, which is all I can think that any conscious entity can ever hope to do.
Just look at the adoption rates on IPv6. No one is going to touch a new protocol at this stage. Its not even clear that this is needed. Point me at a specific TCP pain point that is specifically and obviously reducing internet adoption...any takers?
YAWN Protocol --> Yet Another Wonderful New Protocol
ASCII is still around, despite its numerous shortcomings. There's this small thing called "backward compatibility" that people/consumers seem to love, for some reason. Well, same thing for TCP/IP. Even IPv6 has trouble taking off in the general public, despite being essentially just a small change in the format, so never mind the YAWN Protocol this article is about...
"A door is what a dog is perpetually on the wrong side of" - Ogden Nash
Tried their "Rateless Copy" utility, transferring a 5.8 mb binary file from my web server in Texas to my local connection in Toronto.
With Rateless Copy: time between 31-41 seconds, average of 200k/s, the resulting file is corrupted. Tried it again to ensure, same result.
Without rateless copy (http file download) 8 seconds, average of 490k/s, the resulting file works fine as expected.
Sorry, but I don't think it's all that great.
Why not SCTP ? See RFC 2960. Already in the Linux kernel, Kame, (solaris ?) and probably others.
:-/
Intro here
- SCTP can be used in many "modes"
* Provides reliable messaging (like UDP,but reliable)
* Can be used as a stream protocol (like TCP).
* One connection/association can hold multiple streams.
* One-to-many relation for messaging.
* Better at dealing with syn flooding than TCP.
Then again, I guess inveting the wheel is more "fun"
Really, is TCP flawed?
When considering protocols for information transport, it is very important to be absolutely sure what assumptions you are making. There are a number of non-independent factors which influence the suitability (and hence efficiency) of network protocols to application demands. Bandwidth, for example, is related to but doesn't define the statistical distribution of latencies; maximum packet rate and their relationship to packet size. The channel error rate (and statistical distributions of packet failures) are again linked to fragmentation and concatenation of transmitted datagrams - and this in turn affects latencies when considering "reliable" transport protocols. Routing policy and symmetry of physical links introduces yet more tradeoffs which need to be considered - not to mention the potential problems evaluating if the burden of protocol computations outweighs the advantage of an improved strategy for a given physical link. (And I'm not even going to mention security!) When considering protocols the most important thing to consider is the model they assume of the communications infrastructure on which they are to be deployed. TCP is likely the optimal solution given the assumptions TCP makes... if you change those assumptions to more closely fit a particular network infrastructure you will likely get better performance only on that infrastructure, but far worse performance where your new assumptions do not hold. I used to be interested in the idea of dynamically synthesizing protocols to best suit the actual physical links in a heterogeneous network... however my ideas were met with extreme disinterest; I felt my critics demanded I present a protocol which beats TCP under TCP's assumptions - and no amount of hand-waving and explanation would convince them this was a silly request. I still think the idea has merit - but having wasted 3 years of my life trying to push it uphill, I've found other interesting (and far more productive) avenues to pursue.
While there are a number of issues with TCP, I think it would be much better in the long run to work on fixing TCP rather than replace it. That way all the existing apps can take advantage of the fixes.
One thing that bothers me is I see ISPs applying policing to their subscriber's bandwidth. Policing is quite unfriendly to TCP, unlike, say, shaping. With policing, a router decides either to pass, drop, or mark a packet based on if it exceeds certain bandwidth constraints. Shaping, on the other hand, will buffer packets and introduce additional latency, thus helping TCP find the sweet spot. Of course shaping will also drop, since nobody provides infinite buffer space.
TCP is relatively easy to extend. There are still some free flag bits and additional fields can be added to the TCP header if needed.
-Aaron
This post is encrypted twice with ROT-13. Documenting or attempting to crack this encryption is illegal.
SCTP is indeed interesting. I've tangled with it when playing with SIGTRAN, i.e. SS7 over IP. The nightmares ceased a while ago. :-)
One of the more interesting special-purpose protocols I've ever messed with was the PACSAT Broadcast protocol, used for downloading files from satellites.
It makes the assumption that downloaded files are of general interest, so it broadcasts them. Which files it broadcasts is in response to requests from ground stations. Ground stations can also request fills, pieces of files they haven't received yet. The protocols have hooks to allow files to be downloaded in pieces, over several orbits, since individual passes rarely exceed 20 minutes, and the downlink bit rate isn't all that high. This all runs over a protocol analogous to UDP.
You can get files by just listening, if you want to.
...laura
(NOT) Everyone knows that TCP has problems and for many years people have been developing transport protocols that enhance or replace TCP.
These guys haven't invented anything new. There are many flavours of TCP with different congestion mechanisms and there is a special kind of transport protocol that solves most problems...
I'm talking about SCPS-TP, supported by NASA and it performs very well with high bit-error links (like satellites) and it also copes with high delay. The good thing about SCPS-TP is that it's compatible with TCP, because it basically an extension of TCP.
There is another problem with using UDP based transport protocols... they usually have low priority in routers (probably because you can use UDP for VoIP...)
Fear is the mind-killer.
This work seems to be about two things (which I am not sure I see a strong connection between): lowering transport latency, and using available bandwidth better. The latter has been the subject of many papers in the last few years. There are now several serious proposals of how to fix TCP with respect to long fat pipes. They don't seem to support the idea that retransmissions are harmful. So I'm going to talk about the first issue, transport latency.
The idea of using error-correcting codes (ECC) to eliminate the need for retransmissions is an interesting one. The main benefit is to reduce transport latency (the total time it takes to send data from application A to B). Here is another paper proposes has a similar idea, applied at a different level of the network architecture.
The root problem here is that network loss leads to increases in the transport latency experienced by applications. In TCP, the latency increases because TCP will resend data that is lost. That means at least one extra round-trip-time per retransmission. This "Rateless TCP" approach uses ECC so that the lost data can be recovered from other packets that were not dropped. In this way, the time to retransmit packets may not be needed. I say may, because there will be a loss rate threshold which will exceed the capability of the ECC, and retransmission will become necessary to ensure reliability. But, as long as the loss rate is below the threshold, then retransmissions will not be necessary. Note that the more "resilient" you make the ECC (meaning supporting a higher loss threshold), the more work will be needed at the ends. So you are not eliminating latency due to packet loss, you are simply moving it away from packet retransmission into the process of ECC. However, if you've got good ECC, the total latency will go down.
The ECC approach may be a nice middle ground. But, it the ultimate solution to minimize latency is probably through a combination of active queue management (AQM) and early congestion notification (ECN). Unlike ECC, this approach really would aim to eliminate packet loss in the network due to congestion, and therefore completely eliminate the associated latency. Either ECC or regular TCP would benefit. In a controlled testbed using AQM and ECN, I've completely saturated a network with gigabits of traffic, consisting of thousands of flows, and had virtually no packet loss.
It should also be noted that retransmission is NOT the dominant source of transport latency in of TCP. I am a co-author on a paper that shows another way (other than eliminating retransmission) to greatly reduce the transport latency of TCP. The basic idea is that the send-side socket buffer turns out to be the dominant source of latency (data sits in the kernel socket buffer waiting for transmission). In the above paper, we show how a dynamic socket buffer (one that tracks the congestion window) can dramatically reduce the transport latency of TCP. We allow applications to select this behaviour through a TCP_MINBUF socket option.
-- Buck
This doesn't provide anything like what TCP provides, namely a connection between two network nodes that allows transfer of arbitrary data with guaranteed reliability, with automated congestion control for optimized use of available network resources.
As far as I can tell (their website could use some more straightforward actual content), this is more like bittorrent, where a file is cut up into blocks, the blocks get distributed across the network, and anyone interested in the file then reconstructs it from available data from all sources, not necessarily having to get the entire file correctly from a single source. Only it does it more efficiently than bittorrent.
The two protocols target very different uses. TCP excels in interactive use, where the data is sent as it is generated, and no error is tolerable in the single sender-to-receiver link. Bittorrent (and other distributed network protocols) target batch jobs, where throughput is more important than reliability (because reliability can be reconstructed on the client through clever hashing schemes), and where responsiveness is entirely irrelevant.
So, this could not possibly replace TCP, since it does not do what TCP is most useful for. At the same time, the criticisms aimed at TCP by the rateless designers are valid, but well known, since TCP is indeed poorly suited for high-volume high-throughput high-delay transmissions of prepackaged data.
Still, good job to them for trying to come up with better protocols for niche or not-so-niche markets. I wish them all the best.
How does it work? Well, it's layered over Rateless Internet, in which (as we all know) packets do not have to be resent. So it carefully loses all packets and relies on Rateless Internet to make sure they arrive safely at the other side and do not have to be resent. Because no packets need to make it from A to B, you don't need any network hardware, and data can be sent just as fast as your machine can drop packets.
Guess I'd better apply for a patent...
That's for just them. What if all hosts on the entire Internet were by design stuffing packets at a 3-5% error rate? Meltdown, that's what. Their "real-life" measurements do not scale, suffering from the usual assumed linearity of new designs for complex systems.
Sometimes people fall in love with their new ideas, thinking that the rest of the world missed something obvious.
sigs, as if you care.
So others can have fun slashdotting other technologies, here are some websites. There are probably others, but this should keep those who do really want to move away from TCP happy.
It's a small world and it smells funny; I'd buy another if it wasn't for the money; Take back what I paid (SoM)
ecip.com I call it Error Correcting IP, and used it to stream live video from Sri Lanka in 1997 with Arthur C. Clarke Hal's Birthday
it was a 64K shared line with 90% packet loss, I received 60Kbps for the video stream. ( I have the video to prove it )
We even filled preliminary patents on this back in 1996 but they were never followed through with.
Luigi Rizzo (now head of the FreeBSD project)also did some excellent work on this also. http://info.iet.unipi.it/~luigi/fec.html
He calls it Erasure codes.
Which is more accurate since UDP doesn't have errors, it either come across 99.999% perfect or not at all.
So there is more information then in an error situation where ever bit is questionable.
What this means almost 1/2 the hamming distance in the codes in needed to correct an errasure verses and error.
Turns out the Error/Erasure correcting scheme it critical and not obvious. I spent almost 5 years working on this part time before it started making some real breakthroughs.
My original system was designed for 25% packet loss (not uncommon in 1996).
In the inital idea we added 1 exored packet for every three data packets, but at 25% packet loss, it turns out that it didn't increase reliablity at all! Working this out with probablities was a major eye opener!
Even when you work the problem out you realize you will still need some retransmissions to make up for lost packets, there is no possible solutions without this.
I have been trying to find people to help opensource this since I have working far too hard just to survive since 2000 to even consider taking on another task.
Anyone interested in my research and carring this forward please see my site and contact me.
John L. Sokol
I am always doing that which I can not do, in order that I may learn how to do it. - Pablo Picasso
You get datagrams like in UDP but they are sent re liably.... or not depending on what you want. I believe this has been quietly implemented in CISCO routers and Unix OS vendors network stacks over the past few years.
ayershome.org/users/eric
- They have a "TCP-friendliness" option that varies the transmission rate in a way that TCP windowing can probably cooperate with, so you can set the rate knobs to something less than full blast,
- but nothing they've documented appears to address the problem of multiple users of this application trying to use a transmission path at the same time, and
- they also don't document anything that does path rate discovery - so it may work fine if you've got a couple of small pipes feeding a fat network, but if you've got a fat pipe on the sending end and a skinny pipe on the receiving end, they don't document anything that figures out what rate is safe to transmit at.
They also don't document when you would want to use this and when you would want to use TCP and when you would want to use this on top of TCP.Bill Stewart
New Fast-Compression-only CPR http://preview.tinyurl.com/dy575ks
TCP uses packet loss (NOT round trip times - where did that come from?) to signal congestion, and thus to implement congestion control. This does not work well in a typical wireless (802.11X, 802.16, etc) environment, where packet losses are to be expected. TCP also does not co-exist well with lots of UDP streaming.
So, there IS a need for a TCP replacement. However, the one being developed in the IETF is DCCP. Basically, the idea is separating congestion control and packet loss replacement.
Maymounkov and company say that they are preparing RFC's (which implies that they intend to submit this to the ITEF), but have not yet done so. So, maybe if they do, and if they offer an unencumbered license, their technology could be used, but it is way too soon to tell.
I don't know what their reasoning is, but both their claims about TCP seem incorrect.
1. TCP does not use round trip time to calculate any "congestion levels." It increases the connection rate until packets get dropped, presumably because some router in the middle got overloaded.
2. Packet loss is used as a signal to TCP to slow down because it tried to send too fast. The lost packets are subsequently retransmitted, so TCP can indeed not only tolerate but recover from packet loss. The only real case they have is packet loss due to reasons other than TCP's own aggressive sending rate, such as UDP traffic, wireless links, etc.
Given these concerns, I can't help but think that they are inventing a protocol that works well only if used on a small scale. TCP is designed to back down if it thinks it's sending too fast, and is not really optimal. One can always hack a pair of TCP nodes to not play by the rules and get more than the fair share, but the problem is that that solution wouldn't work if it were adopted network-wide.
Tsunami -- You can't bring a good wave down!
They tell you how they solved the solution, but fail to tell you how much impact this solution has overall.
They're basically stating that now they can flood the connection with packets.
But they've also told you that the packets contain your data in an error correcting encoding. What they don't mention is this:
How much overhead is required by the error correcting encoding?
How many errors can the error correcting encoding handle? (drops 1 packet = ok, drops 400 packets = bad)
How much cpu computation is required to encode and decode the payload?
How is the cpu overhead managed? (how much performance will be lost by context switching, etc.)
So they're just playing the game of distracting people with the best part of thier performance measurement without bothering to mention the performance impact of all of the other trade-offs they admitted to making.
This may be just a wee bit offtopic, but it may be my only chance to ask...
Who remembers HSLink?? If I recall correctly it was an add-on transfer method for Procomm Plus. It allowed two people connected over a modem to simultaneously send files to each other at basically double the normal speed. I remember thinking it had to be a scam, but me and my friends tested it and were able to send double the usual info in whatever time we were used to. (I forget, 10 minutes a meg I think)
How did this work? Were we fooled or was it for reals? Could something like that be applied to dial-up internet connections?
-Don.
Cwm, fjord-bank glyphs vext quiz
TCP doesn't use RTT to 'calculate congestion'.
This is a load of fluff, trying to capitalize on the 'p2p craze'. There are plenty of TCP replacements out there, that actually make sense. As far as TCP not being able to utilize 'today's bandwidth', again...hooey. Gigabit ethernet (when backed by adequate hardware, and taking advantage of jumbo frames) moves a HELL (two orders of magnitude) of a lot more data than your typical home broadband connection...using TCP.
I'd still like to see a good protocol that doesn't require in-order packet receipt in addition to the changes that they mentioned; when transfering large volumes of data, why not?
This is exactly why FTP uses UDP for its data transfer. So use FTP. In the last decade, though, improvements to TCP stacks have mostly mooted this difference. Once upon a time, when one end of a TCP transfer NACK'd a packet, it meant that packet and every packet after it would need to be re-sent (even those which had been sent already). So if you send packets 100, 101, 102 and then get a NACK for 100, you'd have to send 100, 101, 102, 103, etc. But with modern implementations, the receiver would keep packets 101 and 102, so the sender only needs to re-send packet 100, and then proceed to packet 103. This has made TCP much more efficient over lossy networks. Deferred NACKs also deal well with the problem of out-of-order packet arrival, when tardy packets show up before the deferred NACK is transmitted.
I'm very dubious that these folks' protocol can realistically replace TCP; it simply doesn't add any must-have qualities. The only place where I see an advantage is transmitting audio and video, where you don't necessarily want to retransmit lost data, and are willing to put up with minute gaps in the data stream.
A transport protocol that really deserves more attention is TTCP (TCP for Transactions). It abbreviates the TCP connection handshake, which makes it much more efficient / better-performing for very short transactions (like typical HTTP usage).
-- TTK
It is the old-math speak usage of "/" as in "three over four" is "3/4"
So TCP/IP is "Transmission control protocol over internet protocol" or just "TCP over IP".
While you don't see it much, this also inferrs "HTTP/TCP/IP" and so forth.
So just as "voice over IP" doesn't make voice and IP the same, TCP over IP doesn't really relate one to the other.
After all, if you use dialup, the first hop of your connection is typically TCP or UDP over PPP (Point to Point Protocol) and the ISP acts as a gateway from the PPP to the IP semantics.
(I'm guessing the immediate parent knew this, but this reply "hangs best" here in the thread. 8-)
Innocent people shouldn't be forced to pay for inferior software development.
--"Code Complete" Microsoft Press
First order of business is "What is the problem?" This is never really adequately defined, near as I can tell. Some vague references to some "problems" with TCP without any objective data to support the statements. Since a solution is being proposed to a problem, we need some way to measure the problem objectively. How else can you tell if your proposed fix is better or worse than the baseline? Since the problem is never defined in any detail, and certainly never to a point of which you can measure, and the fact that there is no objective data, makes the whole thing totally suspect right from the start. In a nutshell, extraordinariy claims without extraordinary proof
Regardless, there are some nuggets here and there that we can pick apart (from the website):
#1 Rateless Internet is an Internet transport protocol implemented over UDP, meant as a better replacement of TCP. TCP's legacy design carries a number of inefficiencies, the most prominent of which is its inability to utilize most modern links' bandwidth.
#2 This problem stems from the fact that TCP calculates the congestion of the channel based on its round-trip time. The round-trim time, however, reflects not only the congestion level, but also the physical length of the connection. This is precisely why TCP is inherently unable to reach optimal speeds on long high-bandwidth connections.
#3 A secondary, but just as impairing, property of TCP is its inability to tolerate even small amounts (1% - 3%) of packet loss. This additionally forces TCP to work at safe and relatively low transmission speeds with under 1% loss rates. Nevertheless, our extended real-life measurements show that highest throughput is generally achieved at speeds with anywhere between 3% and 5% loss.
Point #1 is a claim without any evidence to back it up. Now most of us may think there is a hint of truth to this from experience. The danger lies in making an assumption that the problem is TCP. It could be a bad implementation of the stack you're using, some kind of packet loss in the path, or you're just trying to send too much data given the conditions of the test (ie, a Pentium 90MHz trying to fill a 10Gb/sec pipe). There's just too many variables that influence the amount of link utilization to make such a generalized statement that "TCP is inherently unable to reach optimal speeds on long high-bandwidth connections". There was a time when two Sun machines sitting next to each other couldn't utilize the entire capacity of a half duplex ethernet, for example. Yet it's not uncommon to have more bandwidth to your home via cable/dsl and make full use of it even on cross country/inter continental links these days without a whole lot of change to the fundamentals of TCP. Therefore, it's unreasonable to conclude that TCP is the cause of any of the problems, perceived or real, considering the history of TCP and it's scaleability over time and total lack of any effort to rule out other causes.
Point #2 is a case of where a little bit of knowledge is much more deadly than none at all. I'd be willing to chalk this one up to a marketing person's copy than any particular design insight by the developers, but it's spooky that it's there. At a fundamental level, queueing theory tells us that the average time for a unit to be processed in a queue is related to the depth of the queue. If there's nothing, or effectively nothing, in the queue when your packet arrives, there is no delay in sending the packet. However, the delay grows as the depth of the queue grows. This shows up as variability in the round trip times of packets, relative to each other. Also known as "jitter". Therefore, no jitter means there are no queueing delays, which means that there isn't much traffic competing for access to the links. High jitter means there are a lot of packets competing for access to a link, and therefore much more likely to be c