Search Results

Search found 149 results on 6 pages for 'qos'.

Page 2/6 | < Previous Page | 1 2 3 4 5 6  | Next Page >

  • Application qos involving priority and bandwidth

    - by Steve Peng
    Our manager wants us to do applicaiton qos which is quite different from the well-known system qos. We have many services of three types, they have priorites, the manager wants to suspend low priority services requests when there are not enough bandwidth for high priority services. But if the high priority services requests decrease, the bandwidth for low priority services should increase and low priority service requests are allowed again. There should be an algorithm involving priority and bandwidth. I don't know how to design the algorithm, is there any example on the internet? Somebody can give suggestion? Thanks. UPDATE All these services are within a same process. We are setting the maximum bandwidth for the three types of services via ports of services via TC (TC is the linux qos tool whose name means traffic control).

    Read the article

  • Is it possible to rate-limit an scp/sftp/rsync/etc transfer from the command-line? ie, manual QoS on

    - by warren
    Specifically, I am looking to rate-limit an scp or sftp session (or other arbitrary network call) in the call itself. For example, let's say I want to copy 100MB to one server, and 1GB to another. I'd like to be able to run both of these at the same time, but maintain a QoS for "normal" computer usage - somewhat similar to how you can rate-limit bittorrent. Is there a way to do this without touching the networking hardware? I'm envisioning something akin to: magic-qos-tool 'scp file user@host:/path/to/file' Or.. scp -rate 40kbps file user@host:/path/to/file

    Read the article

  • Is it possible to rate-limit an scp/sftp/rsync transfer from the command-line? ie, manual QoS on a s

    - by warren
    Specifically, I am looking to rate-limit an scp or sftp session in the call itself. For example, let's say I want to copy 100MB to one server, and 1GB to another. I'd like to be able to run both of these at the same time, but maintain a QoS for "normal" computer usage - somewhat similar to how you can rate-limit bittorrent. Is there a way to do this without touching the networking hardware? I'm envisioning something akin to: magic-qos-tool 'scp file user@host:/path/to/file' Or.. scp -rate 40kbps file user@host:/path/to/file

    Read the article

  • Rsync over NFS with QoS: How to view real transfer speed?

    - by Ian Mackinnon
    We have a bandwidth limit between a Linux server and a NAS, created using 'tc' with an IP filter. When writing to an NFS mount of the NAS, rsync claims a very high transfer speed for each file and then waits a long time before acknowledging that everything has finished. The total time taken is consistent with the QoS limit and the time taken by the same transfer over FTP. Why does the write to the NFS mount report higher transfer speeds than are actually happening over the network? How can I monitor the actual bandwidth of the transfer?

    Read the article

  • Block Skype on Cisco IOS

    - by ensnare
    I'm trying to block skype via policy routing but it's not working ... here's my configuration: class-map match-any block match protocol skype policy-map QoS-Priority-Input class block police 1000000 31250 31250 conform-action drop exceed-action drop violate-action drop policy-map QoS-Priority-Output class block police 1000000 31250 31250 conform-action drop exceed-action drop violate-action drop interface FastEthernet4 description WAN service-policy input QoS-Priority-Input service-policy output QoS-Priority-Output

    Read the article

  • Tomato QoS: Why is some traffic unclassified when there are classifications for it?

    - by Armitage
    Ok, I am trying to tweak my router to give priority to some traffic. My classifications seem to cover just about everything but I still see ~60 to ~80% of the traffic as unclassified: TCP 192.168.1.100 64137 192.168.1.1 80 Unclassified TCP 192.168.1.100 64175 192.168.1.1 80 Unclassified TCP 192.168.1.100 64144 192.168.1.1 443 Unclassified I assume that the 64### ports are just what my WAP uses to send packets inside my home network. But my classifications seems to cover any traffic for destination ports 80 and 443: (partial list) TCP Dst Port: 80,443 High WWW TCP/UDP Dst Port: 1024-65535 Lowest Bulk Traffic Why do I have so much unclassified traffic if I have a classification that should cover it?

    Read the article

  • How do I Implement VLAN Rate Limiting or QOS for a Cisco 2960?

    - by evolvd
    I have a 2960 that I need to limit the uplink port to 50Mbps for 3 vlans and 350Mbps for another vlan. Would the following config achieve that or is this even possible for the 2960? class-map match-any VLAN50-51-52 match vlan 50-52 class-map match-any VLAN53 match vlan 53 policy-map 50MB_RATE_LIMIT class VLAN50-51-52 police 50000000 5000000 exceed-action drop class VLAN53 police 350000000 35000000 exceed-action drop ! interface GigabitEthernet0/23 service-policy output 50MB_RATE_LIMIT service-policy input 50MB_RATE_LIMIT

    Read the article

  • How do I Implement Per VLAN Rate Limiting or QOS for a Cisco 2960?

    - by evolvd
    I have a 2960 that I need to limit the uplink port to 50Mbps for 3 vlans and 350Mbps for another vlan. Would the following config achieve that or is this even possible for the 2960? class-map match-any VLAN50-51-52 match vlan 50-52 class-map match-any VLAN53 match vlan 53 policy-map 50MB_RATE_LIMIT class VLAN50-51-52 police 50000000 5000000 exceed-action drop class VLAN53 police 350000000 35000000 exceed-action drop ! interface GigabitEthernet0/23 service-policy output 50MB_RATE_LIMIT service-policy input 50MB_RATE_LIMIT

    Read the article

  • Mark packets across computers?

    - by eudemo
    I use Transmission on Ubuntu and I'm having this issue, which basically says that QoS is broken because there is no way to limit which outgoing ports uses. I was thinking of doing a dirty and ugly hack and create an interface alias and define QoS based on source address, but was wondering if is there another way. Is it possible to mark the packets on the original machine in some way, using the owner and mark modules of iptables and sending this to the router who does the QoS? From what I understand, mark on iptables only applies to the local machine, so this will not work, but is there another way?

    Read the article

  • Troubleshooting iptables and configuring it to drop the priority of long-term connections

    - by intuited
    I'm somewhat familiar with the general concepts of iptables, and would like to learn it in more detail. I'm hoping that my learning experience can also be useful. The situation: I'm running dd-wrt on my router. Despite its purported QoS skills, I'm still seeing connection latency shoot up hugely whenever there's an ongoing http connection, eg some large download. Under such conditions, it can take 10 seconds or more to load a basic webpage; sometimes the connections are dropped entirely. I've tried adjusting the parameters, dropping the allotted bandwidth for up and download to well under my limit, but nothing seems to work. dd-wrt is configured to use HTB as the QoS algorithm; HFSC, although presented as an option, seems to cause the router to crash, and is rumoured to not actually work on any linux system. I'd like to be able to troubleshoot this issue and hopefully improve the settings that dd-wrt is using, but I'm finding the learning curve a bit overwhelming. For starters I am not sure what HTB actually specifies: is this a set of iptables commands, or do some of those commands specify how HTB is to be used? I would like it to prioritize based on protocol the way that it already supposed to, and in addition I'd like to have it drop the priority of connections which have a high total byte count, say over 400KB. Also tips on utilities that can be run under dd-wrt to get more info on what's going on in there are appreciated. I've tried to get iftop to work but there were issues running curses. I'm leaning towards replacing dd-wrt with openwrt; comments on this strategy are also welcome. I suspect that I would be well advised to get a second router as a standin before trying that. It may be worth noting that my total bandwidth is pretty limited (256Kbit/s).

    Read the article

  • Vlan on astaro 120

    - by Crash893
    (I'm not 100% sure where networking/router questions go this is my best guess) I have a astaro (sophos) white UTM 120 router for work I also have about 11 Voip phones with an externaly hosted pbx (company name = pingtone) Is there any advantage to setting up the phones on a vlan vs making a qos rule that all traffic to my tftp server gets right of way? networking is still a little soft to me Thanks

    Read the article

  • Linux QoS (Skype / BitTorent / SIP / HTTP priority)

    - by Andre
    We are configuring a linux box that will act as internet gateway for an office of 30-50 computers. We are using iptables/HTB for traffic shaping. Is there a way to match traffic on L7 level? It's easy to identify traffic by TCP/UDP ports (like SIP and HTTP). But what if we are dealing with Skype & BitTorent? It was surprise for me that there is no powerful and matured sulution for tasks like this. I found only l7-filter (http://l7-filter.clearfoundation.com/) patch for the Linux kernel, but it's no longer supported (it seems to). Moreover it couldn't be compiled with modern Linux kernels. The only option I found was to use a Cisco router. Are there other ways to identify and shape Skype and Bittorent traffic?

    Read the article

  • How to get better video quality in Lync?

    - by sinned
    I want to use a ConferenceCam from Logitech to stream talks live via Lync. When I view the RAW webcam image via VLC, the quality is very good (but the latency is high because of buffering). However, when I stream it using Lync, the video gets blurry. Is there a way to ensure QoS in Lync or otherwise improve the video quality to (near-)native? I would rather have some dropped frames than a lower resolution where I can't read the slides. In my setup, I use Lync with an Office365-E3 contract, so I have no Lync-Server in my network. I thought about replacing Lync completely with VLC, but I first want to try Lync because VLC will probably cause firewall issues. Also, I haven't looked up the VLC parameters for less buffering, faster encoding, a bit lower resolution (natively it's more than HD) and streaming.

    Read the article

  • Deprioritize BitTorrent traffic

    - by Steven Xu
    I'm sure the question has been asked before, but I can't seem to find it for myself; my Google-fu eludes me. My router, the Linksys E2000, does a decent job at being reasonable about prioritizing some sorts of traffic above BitTorrent traffic (there isn't too much interruption to port 80, 443, or 22 traffic, the ones I use most often). But other ports get pummeled. For instance, 3000 (which I use for local Rails testing) becomes almost entirely non-functioning. Xbox Live traffic (not sure about the ports, but they are in the 1000 range) doesn't do well either. So I'm wondering how to ensure that XBL and local Rails testing maintain strong service while BitTorrent is going. Is it enough that I turn up the QoS on their associated ports to high? It doesn't seem to be as effective as when BitTorrent isn't running at all (I don't know if there's a way to deprioritize BitTorrent traffic).

    Read the article

  • TCP and fair bandwidth sharing

    - by lxgr
    The congestion control algorithm(s) of TCP seem to distribute the available bandwidth fairly between individual TCP flows. Is there some way to enable (or more precisely, enforce) fair bandwidth sharing on a per-host instead of a per-flow basis on a router? There should not be an (easy) way for a user to gain a disproportional bandwidth share by using multiple concurrent TCP flows (the way some download managers and most P2P clients do). I'm currently running a DD-WRT router to share a residential DSL line, and currently it's possible to (inadvertently or maliciously) hog most of the bandwidth by using multiple concurrent connections, which affecty VoIP conversations badly. I've played with the QoS settings a bit, but I'm not sure how to enable fair bandwidth sharing on a per-IP basis (per-service is not an option, as most of the flows are HTTP).

    Read the article

  • How to configure router to give XBox top priority / most bandwidth?

    - by MrSparky
    Hi all - Newbie here so go easy... and apologies in advance if I blow the community etiquette / rules! Here's what I'm trying to do: I have a "D-Link DIR-655 Extreme N" wireless router and an XBox 360 w/ the old-style wireless connection thing (usb attachment)... I want to configure my router / network to give my XBox as much bandwidth as it wants, whenever it wants. I have tried giving the XBox a unique IP (within my network) and then tweaking the router to treat that IP as a top priority application (using the router's QOS stuff). Problem is whenever I turn off the xbox, I can't connect to the network the next time I start it up. It seems the only reliable setting in the XBox is to use "Automatic" for the IP settings within the Network Configuration area. Supposedly the D-Link ships with default settings that attempt to recognize a game console and give it top priority... but i've not seen good results (lots of stuttering / lag when someone else jumps online, etc). Any suggestions? thanks again!

    Read the article

  • Linux TC / Policy Routing tools

    - by Zoredache
    In addition to a really good firewall Linux has a builtin advanced routing and traffic shaping (lartc). There are many applications (firehol, firestarter, etc) to make the creation of iptables firewall easier, what similar to tools exist to make working with the policy routing and traffic control easy?

    Read the article

  • Sharing Bandwidth and Prioritizing Realtime Traffic via HTB, Which Scenario Works Better?

    - by Mecki
    I would like to add some kind of traffic management to our Internet line. After reading a lot of documentation, I think HFSC is too complicated for me (I don't understand all the curves stuff, I'm afraid I will never get it right), CBQ is not recommend, and basically HTB is the way to go for most people. Our internal network has three "segments" and I'd like to share bandwidth more or less equally between those (at least in the beginning). Further I must prioritize traffic according to at least three kinds of traffic (realtime traffic, standard traffic, and bulk traffic). The bandwidth sharing is not as important as the fact that realtime traffic should always be treated as premium traffic whenever possible, but of course no other traffic class may starve either. The question is, what makes more sense and also guarantees better realtime throughput: Creating one class per segment, each having the same rate (priority doesn't matter for classes that are no leaves according to HTB developer) and each of these classes has three sub-classes (leaves) for the 3 priority levels (with different priorities and different rates). Having one class per priority level on top, each having a different rate (again priority won't matter) and each having 3 sub-classes, one per segment, whereas all 3 in the realtime class have highest prio, lowest prio in the bulk class, and so on. I'll try to make this more clear with the following ASCII art image: Case 1: root --+--> Segment A | +--> High Prio | +--> Normal Prio | +--> Low Prio | +--> Segment B | +--> High Prio | +--> Normal Prio | +--> Low Prio | +--> Segment C +--> High Prio +--> Normal Prio +--> Low Prio Case 2: root --+--> High Prio | +--> Segment A | +--> Segment B | +--> Segment C | +--> Normal Prio | +--> Segment A | +--> Segment B | +--> Segment C | +--> Low Prio +--> Segment A +--> Segment B +--> Segment C Case 1 Seems like the way most people would do it, but unless I don't read the HTB implementation details correctly, Case 2 may offer better prioritizing. The HTB manual says, that if a class has hit its rate, it may borrow from its parent and when borrowing, classes with higher priority always get bandwidth offered first. However, it also says that classes having bandwidth available on a lower tree-level are always preferred to those on a higher tree level, regardless of priority. Let's assume the following situation: Segment C is not sending any traffic. Segment A is only sending realtime traffic, as fast as it can (enough to saturate the link alone) and Segment B is only sending bulk traffic, as fast as it can (again, enough to saturate the full link alone). What will happen? Case 1: Segment A-High Prio and Segment B-Low Prio both have packets to send, since A-High Prio has the higher priority, it will always be scheduled first, till it hits its rate. Now it tries to borrow from Segment A, but since Segment A is on a higher level and Segment B-Low Prio has not yet hit its rate, this class is now served first, till it also hits the rate and wants to borrow from Segment B. Once both have hit their rates, both are on the same level again and now Segment A-High Prio is going to win again, until it hits the rate of Segment A. Now it tries to borrow from root (which has plenty of traffic spare, as Segment C is not using any of its guaranteed traffic), but again, it has to wait for Segment B-Low Prio to also reach the root level. Once that happens, priority is taken into account again and this time Segment A-High Prio will get all the bandwidth left over from Segment C. Case 2: High Prio-Segment A and Low Prio-Segment B both have packets to send, again High Prio-Segment A is going to win as it has the higher priority. Once it hits its rate, it tries to borrow from High Prio, which has bandwidth spare, but being on a higher level, it has to wait for Low Prio-Segment B again to also hit its rate. Once both have hit their rate and both have to borrow, High Prio-Segment A will win again until it hits the rate of the High Prio class. Once that happens, it tries to borrow from root, which has again plenty of bandwidth left (all bandwidth of Normal Prio is unused at the moment), but it has to wait again until Low Prio-Segment B hits the rate limit of the Low Prio class and also tries to borrow from root. Finally both classes try to borrow from root, priority is taken into account, and High Prio-Segment A gets all bandwidth root has left over. Both cases seem sub-optimal, as either way realtime traffic sometimes has to wait for bulk traffic, even though there is plenty of bandwidth left it could borrow. However, in case 2 it seems like the realtime traffic has to wait less than in case 1, since it only has to wait till the bulk traffic rate is hit, which is most likely less than the rate of a whole segment (and in case 1 that is the rate it has to wait for). Or am I totally wrong here? I thought about even simpler setups, using a priority qdisc. But priority queues have the big problem that they cause starvation if they are not somehow limited. Starvation is not acceptable. Of course one can put a TBF (Token Bucket Filter) into each priority class to limit the rate and thus avoid starvation, but when doing so, a single priority class cannot saturate the link on its own any longer, even if all other priority classes are empty, the TBF will prevent that from happening. And this is also sub-optimal, since why wouldn't a class get 100% of the line's bandwidth if no other class needs any of it at the moment? Any comments or ideas regarding this setup? It seems so hard to do using standard tc qdiscs. As a programmer it was such an easy task if I could simply write my own scheduler (which I'm not allowed to do).

    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

  • Rate limiting an internet connection per user

    - by Alister
    I've got a friend who has a "rent-by-room" property and includes internet access as part of this. However some tenants are somewhat hogging the internet (i.e. constantly downloading). I was wondering if anyone knows of a fairly easy way of rate limiting each connection to make the system more equitable. A preferred solution would be a cheap piece of hardware or some sort of Linux "appliance". I would rather not have to get an iptables headache if this is avoidable.

    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

< Previous Page | 1 2 3 4 5 6  | Next Page >