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Technology Stocks : The *NEW* Frank Coluccio Technology Forum -- Ignore unavailable to you. Want to Upgrade?


To: Curtis E. Bemis who wrote (829)9/16/2000 2:56:29 AM
From: Frank A. Coluccio  Read Replies (2) | Respond to of 46821
 
Curtis,

As you know, some of the points presented by the poster that you took exception to, concerning LAN utilization metrics, are among the most hotly debated in all of networking.

Before commenting on the validity of his/her observations, I would first want to know the specific configuration of the network in question, and then have an opportunity to view the application transaction profiles that traversed this LAN. And equally important, I'd want to know how many stations the LAN supported.

One of the things that exacerbates this matter is the confusion caused by some otherwise believable individuals who make their living in the rags (and in texts, and studies), whose names many of us would recognize here in a flash, if I printed them. But I'm not paid to make enemies, so we'll leave their names blank.

Alternately, and depending on who you listen to, some blanks will cite the 30% to 40% utilization threshold to doom. Others will cite something in the area of, typically, 75% to 85% utilization before there is cause to start seriously thinking about implementing additional segments.

Keep in mind, we're talking shared media here, consistent with the study that you cited.

I suspect that each of these guesstimates concerning shared media performance represent a throw back to their own personal experiences, since, what the heck, they saw what they saw with their own two eyes. Some of them, granted, have done legitimate investigation. But here, too, depending on what their experimental models looks like and what they are trying to prove, they each yeild different results. More on this point shortly.

Problem is, very few shared Ethernets of the past -- of any appreciable size -- were configured exactly alike, nor did they perform alike. The number and complexity of the variables are just too many and too nitty, respectively, to get into here. [Many of those parameters were covered very nicely in the study you cited in the uplink, for anyone wanting to know what they are.]

Further exacerbating this matter is the persistent obfuscation surrounding the improvements that have taken place in the past nine years, namely, to switching (as opposed to shared media) and full-duplex modes of operation.

But it is useful to point out here that outside of most of the largest enterprises there are still a good number of half-duplex shared Ethernets in existence today, where folks have not found the need or the resources to upgrade, or where the Staples variety of harmonica hub does them just fine. I know of one school library where the librarian is still marveling over their 10Base2 thinnet installation, viewing it as their validation into the Gore Paradigm.

Your citing the now-popular study by Boggs, Mogul and Kent from 1988, however, is, IMO, equally dubious in debating this matter, when you look at: its intent; its place in history; and, the number of nodes that they used in their model to make their point. And the fact that they admittedly were not seeking to exactly replicate real world conditions.

Also, to bring the "real" numbers and improvements that you cited into proper focus, i.e., full-duplex, switched modes of Ethernet, a more-recent study that demonstrates these improvements would have been more useful. [If you or anyone else here knows of a more recent one of equal credibility, please post.]

Twenty-four stations in the first test by Boggs et al, and then twenty three stations in the second test? What were they thinking? I was at least glad to see that the second test reduced the segment lengths, which accounted for reduced collision resolution times. But they didn't go far enough to explore what takes place in real environments, where real station counts and segment lengths were, at the time, being pushed to their limits in large organizations (sometimes over the acceptable limits), across multiple bridged segments.

One must wonder why they limited their experimental model to 24 nodes. When, during the same year, in 1988, I recall Goldman Sachs and Merril Lynch were each pushing over a thousand nodes on similarly shared, bridged thicknet (10Base5) backbones of the multi-segment configuration design.

Let me cite some additional findings from a couple of more-recent studies that have taken place, w.r.t. and since the 1988 study you pointed to by Boggs, et al.

From "Ethernet: The Definitive Guide," by Charles E. Spurgeon, p. 331, re: additional Ethernet performance testing during the 1990s:

"In 1992, Speros Armyros published a paper showing the results of a new simulator for Ethernet that could accurately duplicate the real-world results reported by the Boggs, Mogul and Kent paper. This simulator made it possible to try out some more stress tests of the Ethernet system.

"These new tests replicated the results of the Boggs, Mogul and Kent paper for the 24 stations. The new tests also showed that under worst-case overload conditions, a single Ethenet channel with over 200 stations continually sending data would behave rather poorly, and access times would rapidly increase. "Access time" is the time it takes for a station to transmit a packet onto the channel, including any delays caused by collisions and by multiple packtes backing up in the stations buffers due to congestion of the channel.

"Further analysis of an Ethernet channel using the improved simulator was published by Mart Molle in 1994. Molle's analysis showed that the Ethernet binary exponential backoff algorithm was stable under conditions of constant overload on Ethernet channels with station populations under 200. However, once the set of stations increased much beyond 200, the algorithm begins to respond poorly. In this situation, the access time delays encountered when sending packets can become extremely unpredictable, with some packets encountering rather large delays."

<deleted: section on load definitions>

"The lessons learned in these studies make it clear that users trying to get work done on a constantly overloaded LAN system will perceive unacceptable delays in their network service. Although constant high network loads may feel like a network "collapse" to the users, the network system itself is in fact still working as designed; it's just that the load is too high to rapidly accommodate all of the people who wish to use the network. The delays caused by congestion in shared communication channels are somewhat like the delays caused by congested highways during commute times. Despite the fact that you are forced to endure long delays due to a traffic overload of commuters, the highway system isn't broken; it's just too busy. An overloaded Ethernet channel has the same problem."


Well, I have some nit issues with this last paragraph, but in the broader sense it's okay.

Having copied the above from the text, keep in mind that the author is referring to a shared bus-topology network (as did Boggs, et al) of a size much larger than the one cited by Boggs, et al.

On a personal note, I've seen Ethernets fall into dragass mode, too, with my own two eyes, in our own offices when multiple databases were being rebuilt during periods of heavy CADD activity. Yes, you get to learn what's going on when all goes silent. Earlier, we even saw this occur when using a 4/16 Mb/s Token Ring ;-)

Net-Net: Bandwidth hogs will kill ya every time, theoretical predictions to the contrary, or not.

Comments and corrections welcome.

FAC



To: Curtis E. Bemis who wrote (829)9/16/2000 6:50:37 AM
From: justone  Read Replies (1) | Respond to of 46821
 
Curtis:

Thank you for your referenced article.

It bothers me to see myths propagated. It also appears as though your
knowledge of ethernet (IEEE802.3) technologies ended well over a decade ago.
There are many definitive works that address the nonsense you espoused and the
best known is by Boggs, Mogul and Kent.
research.compaq.com


The article you quote is from 1988, and a production of Digital Equipment
Corporation. A lot of knowledge about traffic is still valid, it seems, even over a
decade later!

Also, many of the sources state that 10Gb Ethernet is basically 1Mb Ethernet, just
faster, so old knowledge still applies. I admit, haven't looked at MAC/PHY/PMD
deeply enough to disagree with these authorities.

I haven't perhaps been clear enough: I'm trying to puzzle out the idea of
multi-access
networks, with many hosts and many clients not managed by a
central group, and understand the traffic issues related to mixing voice demands
(real-time requirements, with fixed small packets) with other media demands such as
data (variable sized packets, and bursty, tolerates delays) or or steam video (large
packets, real-time). This is the traffic model for the famous multi-media broadband
internet (IP based) network, which many think is the next generation network.

I am very happy to thank you for your article, because it gives practical experience
on why collision detection won't do well under multi-access multi host multi-client
conditions, and states that clearly, from the point of theory and simulation. The
lesson's learned section (4.1) give the following recommendations:

". Don't install long cables: to cover a large area, break up the cable with bridges or
gateways (routers), not repeaters.
. Don't put too many hosts on one cable: use gateways to break the network into
communities of interest, trading higher delay for inter­community traffic for better
intra­community response time and throughput.
. Implement the protocol correctly: proper collision detection and binary
exponential backoff in interface or host software is essential to good performance.
. Use the largest possible packet size: this keeps the packet count down, reducing
the likelihood of collision and not incidentally reducing overheads internal to hosts.
. Don't mix serious real­time and serious bulk­data applications: it is not possible
to simultaneously guarantee the lowest delay and the highest throughput (although
for moderate requirements both kinds of applications coexist well)."

Basically, they say to get good performance you must use a MANAGED IP
network, selecting your hosts and client mix with care, setting all parameters in the
network for all clients and servers "properly", and not mix traffic, particularly real
time and large and small packet (VOIP anyone?). This is the opposite of web-world
and internet land, and when you mix demand types. The only recourse other than
management is to heavily over dimension (5-10:1 perhaps?).

I also note the study didn't seem that 'real world' to me (more test lab) and the
number of server and clients is nothing like a web-internet, and the traffic mix was
well regulated.

It might help to keep in mind that even now at 10 Mbps, most
ether implementations are point-to-point, attached to a switch, i.e.. a two station
ethernet, and often full-duplex. This is especially true at 100Mbps and 1GE and will
definitely be that at 10GE.


Well, yes, and I prefer distributed and mesh network topologies to rings myself for
many reasons, but that only confuses our ability to model networks, and thus
manage them. The article recommends that you design your ethernet network with
care, allocating hosts and clients in community of interest. While this may be another
a good argument to buy Akami today (and I think it is), it is also true that most of the
internet can not be managed that way, particularly when you add nomadic computing
to the already open/chatoic/free mix of server hosts and clients.

I really don't like collision detection since noticing when someone else is transmitting
at the same time can take a long time, and this causes long latency, particularly in
WAN fiber multi-mile network, where you must listen long enought for the detection
message to travel the length of the network twice. With high degree of collisions,
and with random back off and re-transmit, this causes bad jitters problems and sinks
your chances of using voice over IP.

Now some these problems can be solved with massive over-bandwidh allocation- I
acknowledge that. And, of course, theses are pre-RSVP and IPv6 analysis- but
when I last looked at RSVP that didn't solve the multi-access multi-host multi-client
problem either: I guess we need to find a paper ls with RSVP real life experience
before passing final judgement. And IPv6, with larger headers, makes small packets
like voice even less efficient.

So let me take a higher level approach.

In terms of control, a useful taxonomy is to divide networks into Fixed Control,
Collision Control, and Dynamic Control. If you analyze the control types of
networks in terms of lost resources, you can have 1) wasted or idle resources, 2)
resources lost due to collision, or 3) control overhead. Fixed networks (leased lines,
TDMA, FDMA, for example) waste resources by having a lot of idle time but are
easy to implement. I note that collision detection (ethernet, for example) networks
don't waste resources, but lose them to collision, and are easy to implement.
Dynamic allocation (switched TDM networks, for example) allocate resources on
demand, and but are more complex to implement and may have higher overheads
(which is one reason why SS7, an overlay network for TDM circuit control was
invented).

My belief is that the best protocols of the future will use dynamic allocation
schemes. These seem more interesting future areas to look at than ethernet. I also
note as investors we should be trying to predict the future, not wait to see if
something deployed succeeds; by then it is too late to get in early.