Search Results

Search found 1797 results on 72 pages for 'bandwidth measuring'.

Page 15/72 | < Previous Page | 11 12 13 14 15 16 17 18 19 20 21 22  | Next Page >

  • In CentOS 4.3 Webmin 1.3000 bandwidth monitoring is eating disk space. How to delete those files?

    - by Silkograph
    I maintain Linux server being used for Mail, Squid and DNS service. Recently I observed that something was eating server disk space. But at last, today I caught the culprit which was consuming the disk by storing large number of files. On this server, Webmin 1.300 is installed. We use Squid proxy and Sarg to monitor Internet access. I always manually clear Sarg generated files under /var/www/html/squid for last few years. But I never realized that Webmin is also storing some kind of bandwidth log files in its' directory structure. I have noticed that under /etc/webmin/bandwidth/hours it has stored more thousands of files since year 2007 totaling about 17 GB. We have used 40 GB HDD for this server machine. My question is how can I delete those (/etc/webmin/bandwidth/hours) files safely?

    Read the article

  • Implications of using many USB web cameras

    - by Martin
    I'm looking into connecting multiple low resolution USB webcams to a single computer. What implications might this have on performance? How does, for example, four 320x240 cameras fare against a single 640x480 camera? I'm not well versed in the architecture of the USB interface, what are the performance caveats? By performance I mean how would it affect the time to read the image data from multiple cameras compared to a single one.

    Read the article

  • php curl upload speed

    - by zanatic
    hey i made a script that uploads files to different servers using php curl and the problem is that it eats all my upload bandwith and my apache does not respond as i would like. the call is made from the php-cli and can not use any apache bandwith limiting bandwith thanks in advance

    Read the article

  • How many bits can be transfered through Ethernet at each time?

    - by Bobb
    I am writing a networking application. It has some unxpected lags. I need to calculate some figures but I cant find an information - how many bits can be transferes through Ethernet connection at each tick. I know that the resulting transfer rate is 100Mbps/1Gbps. But ethernet should use hardware ticks to sync both ends I suppose. So it moves data in ticks. So the question is how many ticks per second or how many bits per one tick used in ethernet. The actual connection is 100 Mbps full-duplex.

    Read the article

  • How to measure the time taken by C# NetworkStream.Read?

    - by publicENEMY
    I want to measure time taken for client to receive data over tcp using c#. Im using NetworkStream.Read to read 100 megabits of data that are sent using NetworkStream.Write. I set the buffer to the same size of data, so there no buffer underrun problem etc. Generally it looks like this. Stopwatch sw = new Stopwatch(); sw.Start(); stream.Read(bytes, 0, bytes.Length); sw.Stop(); The problem is, there is a possibility where the sender hasnt actually sent the data but the stopwatch is already running. how can i accurately measure the time taken to receive the data? i did try to use the time lapse of the remote pc stream.Write, but the time it took to write is extremely small. by the way, is the stopwatch is the most accurate tool for this task?

    Read the article

  • Measure data transfer rate over tcp using c#

    - by publicENEMY
    i want to measure current download speed. im sending huge file over tcp. how can i capture the transfer rate every second? if i use IPv4InterfaceStatistics or similar method, instead of capturing the file transfer rate, i capture the device transfer rate. the problem with capturing device transfer rate is that it captures all ongoing data through the network device instead of the single file that i transfer. how can i capture the file transfer rate? im using c#.

    Read the article

  • Forefront TMG 2010: Can you monitor realtime TCP connections and bandwidth on a per-user basis?

    - by user65235
    I'm just starting a trial of ForeFront TMG to use as a proxy server. I know I can get a real time activity monitor and filter on a per user basis, but would like to be able to get a real time activity monitor of all users that I can then sort by bandwidth consumed (enabling me to get a view on who the bandwidth hogs are). Does anyone know if this is possible in Forefront TMG or if a third party product is required? Thanks. JR

    Read the article

  • Forefront TMG: Can you monitor realtime TCP connections and bandwidth on a per-user basis?

    - by user65235
    I'm just starting a trial of ForeFront TMG to use as a proxy server. I know I can get a real time activity monitor and filter on a per user basis, but would like to be able to get a real time activity monitor of all users that I can then sort by bandwidth consumed (enabling me to get a view on who the bandwidth hogs are). Does anyone know if this is possible in Forefront TMG or if a third party product is required? Thanks. JR

    Read the article

  • How much server bandwidth does an average RTS game require per month?

    - by Nat Weiss
    My friend and I are going to write a multiplayer, multiplatform RTS game and are currently analyzing the costs of going with a client-server architecture. The game will have a small map with mostly characters, not buildings (think of DotA or League of Legends). The authoritative game logic will run on the server and message packet sizes will be highly optimized. We'd like to know approximately how much server bandwidth our proposed RTS game would use on a monthly basis, considering these theoretical constants: 100 concurrent users maximum 8 players maximum per game 10 ticks per second Bonus: If you can tell us approximately how much server RAM this kind of game would use that would also help a great deal. Thanks in advance.

    Read the article

  • Does anyone really understand how HFSC scheduling in Linux/BSD works?

    - by Mecki
    I read the original SIGCOMM '97 PostScript paper about HFSC, it is very technically, but I understand the basic concept. Instead of giving a linear service curve (as with pretty much every other scheduling algorithm), you can specify a convex or concave service curve and thus it is possible to decouple bandwidth and delay. However, even though this paper mentions to kind of scheduling algorithms being used (real-time and link-share), it always only mentions ONE curve per scheduling class (the decoupling is done by specifying this curve, only one curve is needed for that). Now HFSC has been implemented for BSD (OpenBSD, FreeBSD, etc.) using the ALTQ scheduling framework and it has been implemented Linux using the TC scheduling framework (part of iproute2). Both implementations added two additional service curves, that were NOT in the original paper! A real-time service curve and an upper-limit service curve. Again, please note that the original paper mentions two scheduling algorithms (real-time and link-share), but in that paper both work with one single service curve. There never have been two independent service curves for either one as you currently find in BSD and Linux. Even worse, some version of ALTQ seems to add an additional queue priority to HSFC (there is no such thing as priority in the original paper either). I found several BSD HowTo's mentioning this priority setting (even though the man page of the latest ALTQ release knows no such parameter for HSFC, so officially it does not even exist). This all makes the HFSC scheduling even more complex than the algorithm described in the original paper and there are tons of tutorials on the Internet that often contradict each other, one claiming the opposite of the other one. This is probably the main reason why nobody really seems to understand how HFSC scheduling really works. Before I can ask my questions, we need a sample setup of some kind. I'll use a very simple one as seen in the image below: Here are some questions I cannot answer because the tutorials contradict each other: What for do I need a real-time curve at all? Assuming A1, A2, B1, B2 are all 128 kbit/s link-share (no real-time curve for either one), then each of those will get 128 kbit/s if the root has 512 kbit/s to distribute (and A and B are both 256 kbit/s of course), right? Why would I additionally give A1 and B1 a real-time curve with 128 kbit/s? What would this be good for? To give those two a higher priority? According to original paper I can give them a higher priority by using a curve, that's what HFSC is all about after all. By giving both classes a curve of [256kbit/s 20ms 128kbit/s] both have twice the priority than A2 and B2 automatically (still only getting 128 kbit/s on average) Does the real-time bandwidth count towards the link-share bandwidth? E.g. if A1 and B1 both only have 64kbit/s real-time and 64kbit/s link-share bandwidth, does that mean once they are served 64kbit/s via real-time, their link-share requirement is satisfied as well (they might get excess bandwidth, but lets ignore that for a second) or does that mean they get another 64 kbit/s via link-share? So does each class has a bandwidth "requirement" of real-time plus link-share? Or does a class only have a higher requirement than the real-time curve if the link-share curve is higher than the real-time curve (current link-share requirement equals specified link-share requirement minus real-time bandwidth already provided to this class)? Is upper limit curve applied to real-time as well, only to link-share, or maybe to both? Some tutorials say one way, some say the other way. Some even claim upper-limit is the maximum for real-time bandwidth + link-share bandwidth? What is the truth? Assuming A2 and B2 are both 128 kbit/s, does it make any difference if A1 and B1 are 128 kbit/s link-share only, or 64 kbit/s real-time and 128 kbit/s link-share, and if so, what difference? If I use the seperate real-time curve to increase priorities of classes, why would I need "curves" at all? Why is not real-time a flat value and link-share also a flat value? Why are both curves? The need for curves is clear in the original paper, because there is only one attribute of that kind per class. But now, having three attributes (real-time, link-share, and upper-limit) what for do I still need curves on each one? Why would I want the curves shape (not average bandwidth, but their slopes) to be different for real-time and link-share traffic? According to the little documentation available, real-time curve values are totally ignored for inner classes (class A and B), they are only applied to leaf classes (A1, A2, B1, B2). If that is true, why does the ALTQ HFSC sample configuration (search for 3.3 Sample configuration) set real-time curves on inner classes and claims that those set the guaranteed rate of those inner classes? Isn't that completely pointless? (note: pshare sets the link-share curve in ALTQ and grate the real-time curve; you can see this in the paragraph above the sample configuration). Some tutorials say the sum of all real-time curves may not be higher than 80% of the line speed, others say it must not be higher than 70% of the line speed. Which one is right or are they maybe both wrong? One tutorial said you shall forget all the theory. No matter how things really work (schedulers and bandwidth distribution), imagine the three curves according to the following "simplified mind model": real-time is the guaranteed bandwidth that this class will always get. link-share is the bandwidth that this class wants to become fully satisfied, but satisfaction cannot be guaranteed. In case there is excess bandwidth, the class might even get offered more bandwidth than necessary to become satisfied, but it may never use more than upper-limit says. For all this to work, the sum of all real-time bandwidths may not be above xx% of the line speed (see question above, the percentage varies). Question: Is this more or less accurate or a total misunderstanding of HSFC? And if assumption above is really accurate, where is prioritization in that model? E.g. every class might have a real-time bandwidth (guaranteed), a link-share bandwidth (not guaranteed) and an maybe an upper-limit, but still some classes have higher priority needs than other classes. In that case I must still prioritize somehow, even among real-time traffic of those classes. Would I prioritize by the slope of the curves? And if so, which curve? The real-time curve? The link-share curve? The upper-limit curve? All of them? Would I give all of them the same slope or each a different one and how to find out the right slope? I still haven't lost hope that there exists at least a hand full of people in this world that really understood HFSC and are able to answer all these questions accurately. And doing so without contradicting each other in the answers would be really nice ;-)

    Read the article

  • Does anyone really understand how HFSC scheduling in Linux/BSD works?

    - by Mecki
    I read the original SIGCOMM '97 PostScript paper about HFSC, it is very technically, but I understand the basic concept. Instead of giving a linear service curve (as with pretty much every other scheduling algorithm), you can specify a convex or concave service curve and thus it is possible to decouple bandwidth and delay. However, even though this paper mentions to kind of scheduling algorithms being used (real-time and link-share), it always only mentions ONE curve per scheduling class (the decoupling is done by specifying this curve, only one curve is needed for that). Now HFSC has been implemented for BSD (OpenBSD, FreeBSD, etc.) using the ALTQ scheduling framework and it has been implemented Linux using the TC scheduling framework (part of iproute2). Both implementations added two additional service curves, that were NOT in the original paper! A real-time service curve and an upper-limit service curve. Again, please note that the original paper mentions two scheduling algorithms (real-time and link-share), but in that paper both work with one single service curve. There never have been two independent service curves for either one as you currently find in BSD and Linux. Even worse, some version of ALTQ seems to add an additional queue priority to HSFC (there is no such thing as priority in the original paper either). I found several BSD HowTo's mentioning this priority setting (even though the man page of the latest ALTQ release knows no such parameter for HSFC, so officially it does not even exist). This all makes the HFSC scheduling even more complex than the algorithm described in the original paper and there are tons of tutorials on the Internet that often contradict each other, one claiming the opposite of the other one. This is probably the main reason why nobody really seems to understand how HFSC scheduling really works. Before I can ask my questions, we need a sample setup of some kind. I'll use a very simple one as seen in the image below: Here are some questions I cannot answer because the tutorials contradict each other: What for do I need a real-time curve at all? Assuming A1, A2, B1, B2 are all 128 kbit/s link-share (no real-time curve for either one), then each of those will get 128 kbit/s if the root has 512 kbit/s to distribute (and A and B are both 256 kbit/s of course), right? Why would I additionally give A1 and B1 a real-time curve with 128 kbit/s? What would this be good for? To give those two a higher priority? According to original paper I can give them a higher priority by using a curve, that's what HFSC is all about after all. By giving both classes a curve of [256kbit/s 20ms 128kbit/s] both have twice the priority than A2 and B2 automatically (still only getting 128 kbit/s on average) Does the real-time bandwidth count towards the link-share bandwidth? E.g. if A1 and B1 both only have 64kbit/s real-time and 64kbit/s link-share bandwidth, does that mean once they are served 64kbit/s via real-time, their link-share requirement is satisfied as well (they might get excess bandwidth, but lets ignore that for a second) or does that mean they get another 64 kbit/s via link-share? So does each class has a bandwidth "requirement" of real-time plus link-share? Or does a class only have a higher requirement than the real-time curve if the link-share curve is higher than the real-time curve (current link-share requirement equals specified link-share requirement minus real-time bandwidth already provided to this class)? Is upper limit curve applied to real-time as well, only to link-share, or maybe to both? Some tutorials say one way, some say the other way. Some even claim upper-limit is the maximum for real-time bandwidth + link-share bandwidth? What is the truth? Assuming A2 and B2 are both 128 kbit/s, does it make any difference if A1 and B1 are 128 kbit/s link-share only, or 64 kbit/s real-time and 128 kbit/s link-share, and if so, what difference? If I use the seperate real-time curve to increase priorities of classes, why would I need "curves" at all? Why is not real-time a flat value and link-share also a flat value? Why are both curves? The need for curves is clear in the original paper, because there is only one attribute of that kind per class. But now, having three attributes (real-time, link-share, and upper-limit) what for do I still need curves on each one? Why would I want the curves shape (not average bandwidth, but their slopes) to be different for real-time and link-share traffic? According to the little documentation available, real-time curve values are totally ignored for inner classes (class A and B), they are only applied to leaf classes (A1, A2, B1, B2). If that is true, why does the ALTQ HFSC sample configuration (search for 3.3 Sample configuration) set real-time curves on inner classes and claims that those set the guaranteed rate of those inner classes? Isn't that completely pointless? (note: pshare sets the link-share curve in ALTQ and grate the real-time curve; you can see this in the paragraph above the sample configuration). Some tutorials say the sum of all real-time curves may not be higher than 80% of the line speed, others say it must not be higher than 70% of the line speed. Which one is right or are they maybe both wrong? One tutorial said you shall forget all the theory. No matter how things really work (schedulers and bandwidth distribution), imagine the three curves according to the following "simplified mind model": real-time is the guaranteed bandwidth that this class will always get. link-share is the bandwidth that this class wants to become fully satisfied, but satisfaction cannot be guaranteed. In case there is excess bandwidth, the class might even get offered more bandwidth than necessary to become satisfied, but it may never use more than upper-limit says. For all this to work, the sum of all real-time bandwidths may not be above xx% of the line speed (see question above, the percentage varies). Question: Is this more or less accurate or a total misunderstanding of HSFC? And if assumption above is really accurate, where is prioritization in that model? E.g. every class might have a real-time bandwidth (guaranteed), a link-share bandwidth (not guaranteed) and an maybe an upper-limit, but still some classes have higher priority needs than other classes. In that case I must still prioritize somehow, even among real-time traffic of those classes. Would I prioritize by the slope of the curves? And if so, which curve? The real-time curve? The link-share curve? The upper-limit curve? All of them? Would I give all of them the same slope or each a different one and how to find out the right slope? I still haven't lost hope that there exists at least a hand full of people in this world that really understood HFSC and are able to answer all these questions accurately. And doing so without contradicting each other in the answers would be really nice ;-)

    Read the article

  • GMail IMAP + Apple Mail / iPhone - "Account exceeded bandwidth limits. (Failure)"

    - by bpapa
    Started seeing this this morning in Apple Mail. I have one of those exclamation point error indicators next to "Inbox", with this error message when I click on it: There may be a problem with the mail server or network. Verify the settings for account “IMAP Account” or try again. The server returned the error: Account exceeded bandwidth limits. (Failure). This is in Snow Leopard. I'm using GMail IMAP, and I am way below the size quota - I've never heard of there even being a bandwidth quota. I'm also not getting mail from the same account to the mail app on my iPhone. EDIT - a month later I'm seeing this, and I'm thinking of just switching Mobile Me. EDIT AGAIN - Making community wiki. I stopped seeing the problem once I updated Snow Leopard to the latest version, but since others continue to see it...

    Read the article

  • Is measuring software project metrics popular in todays industry?

    - by Russ K
    I encountered a developer who wanted some outside advice on their teams project. I found out they're developing a huge software suite for the companies executives, project manager and developers that can calculate metrics automatically and graph them per iteration. As a student from a computer science background I know very little on metrics and their importance, but my questions are: Do most companies have some way, doesn't have to be an elegant program, to measure meaningful metrics? Which metrics, single or combined, help you narrow down your projects scope and estimates? As a person who analyzes metrics, how often do you base decisions off of them? IE. Tests failed per week is increasing drastically? Do you feel that the introduction of studying metrics has helped you understand the project better? Not sure why but the developers project intrigued me and I must know more. If y

    Read the article

  • How would I go about measuring the impact an article has on the internet?

    - by Jimbo Mombasa
    For an application of mine, I analyze the sentiment of articles, using NLTK, to display sentiment trends. But right now all articles weigh the same amount. This does not show a very accurate picture because some articles have a higher impact on the internet than others. For example, a blog post from some unknown blog should not weigh the same amount as an article from the New York Times. How can I determine their impact?

    Read the article

  • How exactly is Google Webmaster Tools measuring "Site Performance"?

    - by Rémi
    I've been working for two months now on improving our response time (mainly server side) on a new forum (a brand new product on a technical point of view) we've launched in Germany a few month ago and I'm a lot surprised by the results I get. I monitor our response time using Apache logs and our own implementation of Boomerang beacon. Using my stats, I can see that our new product responds in about 680 ms where our old product was responding in about 1050 ms. On the other side, Google Webmaster Tool tells us that our pages have an average reponse time of about 1500 ms today where it was 700 three months ago with our old product. I've figured that GWT was taking client side metrics into account so I've added some measures on our Boomerang beacon and everything looks just fine. I've also ran some random pages on ySlow and Google's Page Speed and everything looks better than it was before. We event have a 82% on Google's Page Speed tool which is quite cool for a site with some ads in it :) Lately, we have signed a deal with Akamai to use two of their products : CDN for our static files (we were using another CDN before but it wasn't very effective) and RMA to improve Networks routes. We have also introduced a new agressive cache mecanism to ensure that most of the pages served to crawlers are cached by our memcache grid. After checking my metrics, it seems that this changes have improved from 650ms to about 500ms, which is good (still not great but it is definitly an improvement). But webmaster tools continues to report an increasing average response time where we see it decreasing in the same time. Have you ever had the same kind of wierd behavior on your sites while doing performance improvements ? Do you have any idea how to monitor the same thing Google does with Site Performance in Google Webmaster Tools so that we could improve our site and constantly check if it is what Google wants ? Edit 2011/07/26 : Thanks for your answers guys ! Nevertheless, I was not precise enough. The main issue we have is not with the Site Performance page but with the Crawl Stats one for now. We probably found an issue on our side with some very slow pages (around 3000 ms !!) and we are trying to fix them. I'll keep you posted as soon I'll have some infos. Thanks again !

    Read the article

  • QoS for Cisco Router to Prioritize Voice and Interactive Traffic

    - by TJ Huffington
    I have a Cisco 891W NATing Voice and Data to the internet over a 10mbit/2mbit connection. Voice traffic gets degraded when I upload large files. Pings time out as well. I tried to configure a QoS policy but it's basically not doing anything. Voice traffic still degrades when upload bandwidth gets saturated. Here is my current configruation: class-map match-any QoS-Transactional match protocol ssh match protocol xwindows class-map match-any QoS-Voice match protocol rtp audio class-map match-any QoS-Bulk match protocol secure-nntp match protocol smtp match protocol tftp match protocol ftp class-map match-any QoS-Management match protocol snmp match protocol dns match protocol secure-imap class-map match-any QoS-Inter-Video match protocol rtp video class-map match-any QoS-Voice-Control match access-group name Voice-Control policy-map QoS-Priority-Output class QoS-Voice priority percent 25 set dscp ef class QoS-Inter-Video bandwidth remaining percent 10 set dscp af41 class QoS-Transactional bandwidth remaining percent 25 random-detect dscp-based set dscp af21 class QoS-Bulk bandwidth remaining percent 5 random-detect dscp-based set dscp af11 class QoS-Management bandwidth remaining percent 1 set dscp cs2 class QoS-Voice-Control priority percent 5 set dscp ef class class-default fair-queue interface FastEthernet8 bandwidth 1024 bandwidth receive 20480 ip address dhcp ip nat outside ip virtual-reassembly duplex auto speed auto auto discovery qos crypto map mymap max-reserved-bandwidth 80 service-policy output QoS-Priority-Output crypto map mymap 10 ipsec-isakmp set peer 1.2.3.4 default set transform-set ESP-3DES-SHA match address 110 qos pre-classify ! fa8 is my connection to the internet. Voice traffic goes over a VPN ("mymap") to the SIP server. That's why I specified "qos pre-classify" which I believe is the way to classify traffic over the VPN. However even when I ping a public IP while saturating upload bandwidth, the latency is exceptionally high. Is this configuration correct? Are there any suggestions that might make this work for my setup? Thanks in advance.

    Read the article

  • QoS for Cisco Router to Prioritize Voice and Interactive Traffic

    - by TJ Huffington
    I have a Cisco 891W NATing Voice and Data to the internet over a 10mbit/2mbit connection. Voice traffic gets degraded when I upload large files. Pings time out as well. I tried to configure a QoS policy but it's basically not doing anything. Voice traffic still degrades when upload bandwidth gets saturated. Here is my current configruation: class-map match-any QoS-Transactional match protocol ssh match protocol xwindows class-map match-any QoS-Voice match protocol rtp audio class-map match-any QoS-Bulk match protocol secure-nntp match protocol smtp match protocol tftp match protocol ftp class-map match-any QoS-Management match protocol snmp match protocol dns match protocol secure-imap class-map match-any QoS-Inter-Video match protocol rtp video class-map match-any QoS-Voice-Control match access-group name Voice-Control policy-map QoS-Priority-Output class QoS-Voice priority percent 25 set dscp ef class QoS-Inter-Video bandwidth remaining percent 10 set dscp af41 class QoS-Transactional bandwidth remaining percent 25 random-detect dscp-based set dscp af21 class QoS-Bulk bandwidth remaining percent 5 random-detect dscp-based set dscp af11 class QoS-Management bandwidth remaining percent 1 set dscp cs2 class QoS-Voice-Control priority percent 5 set dscp ef class class-default fair-queue interface FastEthernet8 bandwidth 1024 bandwidth receive 20480 ip address dhcp ip nat outside ip virtual-reassembly duplex auto speed auto auto discovery qos crypto map mymap max-reserved-bandwidth 80 service-policy output QoS-Priority-Output crypto map mymap 10 ipsec-isakmp set peer 1.2.3.4 default set transform-set ESP-3DES-SHA match address 110 qos pre-classify ! fa8 is my connection to the internet. Voice traffic goes over a VPN ("mymap") to the SIP server. That's why I specified "qos pre-classify" which I believe is the way to classify traffic over the VPN. However even when I ping a public IP while saturating upload bandwidth, the latency is exceptionally high. Is this configuration correct? Are there any suggestions that might make this work for my setup? Thanks in advance.

    Read the article

  • What we have to measure for measuring server performance If we can't measure the server processing time from client side?

    - by AsadYarKhan
    If we can not measure the server processing time from client side then which attributes will be good to measure in client side for measuring server side performance and What attributes are important ? I know we can get the server response time, latency and Throughput etc,but how do we understand/interpret the result of server side from these attrubutes. How can we analyse that whether my code is taking lots of time,whether Web Server, whether it is because of Server Machine(H/W).how would i know that which thing needs to be upgrade or improve.Please tell me any article or any book something that I need to study or explain here If you can so I can interpret the result of server side using these attributes response time, latency and throughput.You can tell other performance attribute if I need to understand the server result.

    Read the article

  • Apache2: Limit simultaneous requests & throttle bandwidth per IP/client?

    - by xentek
    I want to limit simultaneous requests & throttle bandwidth per IP/Client on a single apache vhost. In other words, I want to ensure that this site, which hosts large media files, doesn't get hammered by someone trying to download everything all at once (just happened the other night). I'd like to limit the outgoing transfer speed overall for this site, as well as limit the number of connections a single IP can make to the server to a sane default (i.e. within normal browser limits for multiple requests so page loads aren't effected too much). Bonus points if I can actually scope it to file types (i.e. leave web files alone, but apply these rules to just the media files). We're running Ubuntu 9.04 on all the servers, and have two apache/php servers being load balanced via Round Robin by a squid proxy server. MySQL is running on its own box as well. We've got plenty of bandwidth to give them, so I don't really want overall caps, but just want to throttle the amount of memory/CPU it takes to serve this site. There other sites on these servers that we don't want to apply these rules too, just want to keep this one from hogging all the resources. Let me know if you need more info! Thanks in advance for your suggestions!

    Read the article

  • Application that will identify percentage of your system disk bandwidth used on a user-application by user-application basis?

    - by Warren P
    I always (subjectively) feel my computer is far too slow (however fast it is), and so I'm always looking for ways to measure and understand what my computer is actually doing, that is making it seem "slow" to me. It has been my observation that my software-developer workload is most often disk-bound (I am waiting for Disk I/O) more than CPU bound. What has made it worse, is that I am using a corporate PC that has in-memory active-scanning anti-virus software that I do not have control over, and also some IT department mandated services that seem to suck up a lot of available hard-disk bandwidth. The best tool I have seen (in Windows 7) is the Resource Monitor which I usually acess from the button in the task Manager. The disk IO page, however, seems to label Disk Activity at a very low level (for example, showing the Volume Shadow Storage, which is flushing information obviously written by something ELSE other than VSS itself, and then writes to Pagefile.sys, which are obviously due to Virtual Memory faults in some application). What I would like to know is if a utility exists that can add up all direct disk input and output by user-level process, or find the process or service that caused VM or VSS activity. In that way, I hope, you could establish a real idea of how much of your computer's precious disk subsystem bandwidth is attributable to a particular application. here's a scenario: MyApp.exe writes 100k/s and reads 100k/s directly. VSS ends up writing another 100k/s. pagefaults caused inside MyApp.exe cause another 100k/s of writes. So the total "cost" of MyApp.exe running, during a period of time (let's say 1 second) is 400k/s, whereas you can only directly observe half of that, in Resource Monitor. Is there a smarter disk-IO watching piece of software I can use?

    Read the article

  • Throughput and why do ISPs sell too much bandwidth?

    - by jonescb
    I hope the question made sense how I worded it. :) I've been wondering, maximum theoretical bandwidth is measured as RWIN/RTT (Window size / round trip time) Source 1 and Souce 2 So if a major city only 100 miles away gives me a ping of 50ms, and I have the default 64kb TCP window size then my maximum throughput will be 12.5Mb/s. Everything further away would give me a higher ping and therefore a lower throughput. Is there any reason to buy something like FiOS with a 50Mb/s or greater connection? Will you ever be able to reach that kind of speed? I know you can increase the TCP window size to increase throughput, but it has to be at both ends which is a deal breaker because you can't control the server. I'm assuming other network protocols like UDP aren't quite as affected by latency as TCP is, but how much of overall network traffic does non-TCP make up vs TCP. Am I just misguided about how throughput works? But if the above is correct, then why should a consumer like me buy way more bandwidth than can be realistically used. Maybe the only reason is for downloading multiple things at once, or one thing from multiple servers/peers?

    Read the article

< Previous Page | 11 12 13 14 15 16 17 18 19 20 21 22  | Next Page >