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The Internet

Maximum Latency for ISPs? 127

Posted by Cliff from the quality-of-service dept.
fluor2 asks: "My ISP is providing me 8mbit ADSL, and my speed is in fact 8mbit (downstream). However, we all know that there is no relation between transfer rate and latency, eg, a high transfer rate and high latency will kill your FPS games. A packet that travels through the sky and up to a satellite is bound to give high latency. Using pathping, I discovered that my ISP provides me with a latency of 22ms before my sent packets are sent out of my ISP's backbone (6 hops). I have a friend that also tried the same, and he got only 10ms before he was out of his ISP's network. I know 22ms is decent, but I still think that it's far too high if one uses IP-phones and similar. What kind of latency can we accept for a normal 8mbit ADSL connection, and isn't it about time that we get more focus on this subject?"
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Maximum Latency for ISPs? More

Maximum Latency for ISPs?

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  • "Normal" 8Mb? (Score:5, Interesting)

    by leonbrooks ( 8043 ) <SentByMSBlast-No ... .brooks.fdns.net> on Wednesday July 30, 2003 @08:49PM (#6576070) Homepage
    I've got 512/128kb and consider it to be luxury. Perth, West Oz.

  • Well... (Score:5, Funny)

    by Anonymous Coward on Wednesday July 30, 2003 @08:50PM (#6576074)
    An 8mbps DSL line...

    Since you're one of the first folks to try out this new tech, I think you need to tell US what to expect.

    How much do you pay for that thing anyways? Just to play games?

    Holy shit. I have trouble putting food on the table and you're worried about your high latency times for an 8mbps connection?
    • The point being....

      If he's paying some crazy sum of money he expects to get what he pays for, right?

      If I'm on dail-up I expect a slow connection. If I'm on a DSL 512kb I expect a 512kb connection. If I pay an arm, a leg, and sold my soul to the devil I expect a speedy connection with low latency.
    • Holy shit. I have trouble putting food on the table and you're worried about your high latency times for an 8mbps connection?
      Have you considered serving trays?
    • I also happen to have a similar connection, although I'm not complaining about my latencies... I have 8Mbps(down) ADSL with NTT in the Osaka area of Japan. I also had the same service in Yokohama until recently. It's very cheap (about $22/month) too. I have a 1.5Mbps(down) ADSL in San Diego with SBC which is decent, but has too many outages for my taste. It's more expensive (about $50/month) Z.

  • Comment removed (Score:5, Informative)

    by account_deleted ( 4530225 ) on Wednesday July 30, 2003 @08:52PM (#6576090)
    Comment removed based on user account deletion

    • Re:CAP or DMT? (Score:2, Informative)

      by ManDude ( 231569 )
      Telcos normally aim for 150ms or lower. It isn't a problem for local, but cross-country + trans-ocean it can be a bit of a problem. When I call the UK you can catch bad switching. One second delay is brutal since it ends up being 2 seconds for return. I have ended a conversation saying "bye" more then thrice :) Anything like 22ms + a few 10s would be ideal.

    • Re:CAP or DMT? (Score:4, Informative)

      by doogles ( 103478 ) on Wednesday July 30, 2003 @10:16PM (#6576547)
      Some codecs don't play nicely with high latencies,

      It's my belief this is a myth.

      VoIP itself cares little about one-way delay, but cares a whole lot about jitter. If I can provide you with a one-way link that has extremely high one-way delay, but I have routers on each end of the link to ensure voice gets queued and transmitted before any other data in the queue, the service will be acceptable. The only piece that may be unreasonable to users is the delay itself; the ITU standard is 150ms of one-way delay (300ms, as you mention, would be a correct "round trip" time assuming delay in each direction was the same).

      High delay can lead to users talking over one another; we're not use to this when we call granny down the street. But for remote locations or international calls, they are use to extremely high delays and so taking their call across an extremely high-delay path (such as satellite, as you mention) results in no net difference for the end-user. Yes, VoIP endpoints will add some of their own delays, but these will be fairly insignificant if you're talking about a 500ms one-way delay budget.

      I have a collegue with a customer who has 100+ sites about Alaska, in VERY rural areas. All their voice calls go across an satellite-transported IP network. Sure, _you_ might have a little trouble getting use to it at first, but the regular users are use to high delays on their calls (much like your cell phone, as you pointed out). In the grand scheme of things, I would argue that as long as the packets arrive jitter-free (meaning there are not huge inter-frame gaps, which would mean there is nothing for the far end codec to decode and play out) the quality of the call itself will be acceptable.

      In the end, though, I think we both agree -- I just don't know that I've ever run in to an environment where delay was the cause of "poor voice quality". Loss and jitter will typically be the root cause for why codec isn't producing the quality you'd expect. Delay is just, well, delay.
    • Just to follow up, VoIP can indeed function very well with higher latencies, including the ranges you mentioned above.

      The most important concern that that the latency remain fairly constant. If you have a consistant 300ms latency, the call will be great. If, however, the latency fluctuates between 100ms and 400ms, that's where call quality can rapidly go down the tubes.

      Consistant latency is the most important factor for streaming traffic.

    • However, most telcos are switching to DMT because I think it's more scalable.

      Dude, you have POWER! If you thought IP over carrier pigeons were more scalable, would they switch to that? ;)

  • That's good.. (Score:3, Interesting)

    by PFAK ( 524350 ) on Wednesday July 30, 2003 @08:52PM (#6576092)
    I get 15ms to my ADSL modem, and I used to get 33ms. You are getting pretty good pings if it's still in your ISP, except about 40ms in your ISP.

    I don't see whats wrong with what you are getting, maybe you are whining just a little bit too much about what you are getting.

    Heck, I'd like 8mbps down on my ADSL. I'm stuck with 1.53mbps/640kbps.

    Oh well. There is nothing wrong with what you get..

    • Re:That's good.. (Score:4, Funny)

      by Transcendent ( 204992 ) on Thursday July 31, 2003 @11:15AM (#6580209)
      I get 15ms to my ADSL modem

      From your computer to your MODEM??? How many miles of cable are you going through to the modem that sits by your computer??
    • Heck, I'd like 8mbps down on my ADSL. I'm stuck with 1.53mbps/640kbps.

      Oh, to be so lucky to have 1.53/640. I can't even get *cable TV* where I live. I do the happy dance when I'm able to connect at 40K on dial-up.

      (insert "and I was *grateful*" speech here)

      Someone's always got it worse off than you.
    • I just tried it myself, and guess what? I get 22ms too. However I don't have 8mbps DSL.

      We used to get DSL through the local phone company. 100-110ms was not that uncommon during the day. (I kid you not!)

      Interestingly, it takes 3ms to get to the modem. Actually that's not that bad considering about 50ft of cable between this computer and the router/modem in the basement.

  • Spoiled (Score:3, Insightful)

    by medeii ( 472309 ) on Wednesday July 30, 2003 @08:58PM (#6576122)

    isn't it about time that we get more focus on this subject?

    About time, sure. If I could get anything other than no-server cable, I'd be sure to jump on your bandwagon.

    Can we focus on getting decent broadband to everyone first, and THEN start worrying about 12 ms of ping time? Good god, man.

    • Spot on. I have family who have almost never used the net and say hey wow you get that in your computer, how do ya do that, can I? (Streaming video and radio). Then you explain that they live in the UK and thus their exhange is not enabled yet (for ADSL) and even if it was you are probably too far from it to get broadband(512K/s up 256K/s down). All they really want is 128k/s upstream always on! Oh what is ti that I hear "thats that", not "oh we should do something about this," so give a thought to thouse

    • Can we focus on getting decent broadband to everyone first, and THEN start worrying about 12 ms of ping time? Good god, man.

      You'll get it when you get it. It takes approximately one hour to convince yourself that your pre-broadband days were just a distant memory. We're working hard to make sure that one hour after you get broadband that you are still happy.

      But seriously, the pingpath app linked above runs on some weird OS that I don't use. Does anybody know what the equivalent Linux functionality is?
      • traceroute does something similar. It displays latency to all the hops between you and the traceroute parameter.
      • http://www.bitwizard.nl/mtr/

        What is MTR?
        mtr combines the functionality of the 'traceroute' and 'ping' programs in a single network diagnostic tool.
        As mtr starts, it investigates the network connection between the host mtr runs on and a user-specified destination host. After it determines the address of each network hop between the machines, it sends a sequence ICMP ECHO requests to each one to determine the quality of the link to each machine. As it does this, it prints running statistics about each machi
    • (Gaming refers to online FPS gaming, not MMORPG since i know nothing about network requirements for those.)

      Look it actually matters for gaming. In fact upstream matters more than downstream if you are playing quake3. (upstream traffic is about 1.5 times more than downstream).

      If you dont play games, the current development of broadband connectivity is going to help you. Nobody is really considering what games needs, just 'more downstream, more downstream'.

      Gaming needs better upstream & very very low l

  • Boohoo.. (Score:5, Funny)

    by OutRigged ( 573843 ) <rage@o u t r i g g e d . com> on Wednesday July 30, 2003 @09:00PM (#6576135) Homepage
    You've got an 8mbit a second ADSL connection, and you get 22ms pings? Cry me a river.

    Alright, yeah. I'm jealous. :(

    • Re:Boohoo.. (Score:2, Funny)

      by spuke4000 ( 587845 )
      Next on Ask Slashdot:
      • My diamond shoes are too tight, what should I do?
      • All these fifties won't fit in my wallet, can you help?
      • I'm having sex with lingrie model after lingerie model, what am I supposed to do???
    • I had myself a cable internet conenction this year.

      I liked the speed, frequently between 2 and 3 mbps. But, I complained about the lag. My ISP didn't get a complaint phonecall until my GATEWAY became at least 150ms away (instead of the normal 75ms), let alone the internet...

      I found that average internet ping time was 600ms. this was a problem. I was used to a fast cable provider which could offer internet ping times in the 150-250ms range. (and gateway pings in the 20-40ms)

      So please, stop compl

  • 22ms? (Score:5, Funny)

    by alph0ns3 ( 547254 ) on Wednesday July 30, 2003 @09:03PM (#6576153)
    Back then, we had 33,6k modems, with 200ms pings at best, we played quake in software mode in 320x240 at 10 fps, and we were happy!
    • Hey, you think that's latency, consider people forced to use this protocol [ietf.org].
    • " we played quake in software mode in 320x240 at 10 fps, and we were happy!"

      I remember being accused of cheating because I had a 3DFX card.
    • Yup. This post makes me want to dl the Linux QuakeWorld binaries one more time and dust off the old CD...
    • Back then, we had 33,6k modems, with 200ms pings at best, we played quake in software mode in 320x240 at 10 fps, and we were happy!

      You LPB! Us real HPB were 300ms and up!
    • Quake? Why back in my day we used to play Doom using 2400 baud modems over noisy lines. You don't know nothing kid...
    • Well, if we're going to go on about the olden days...

      Why I remember playing Land Of Devistation on my 1200 baud modem. I would get less then 1 FPS! Killing mutant valley girls and avoiding land mines took real skill! And we were HAPPY, dangnabbit! You consarnit whippersnappers and your fancy shmancy 3D games makes my blood boil. Why if I had my electric sword, I show you ALL what REAL gaming is like!

  • Cable pings better than DSL (Score:4, Informative)

    by xWeston ( 577162 ) on Wednesday July 30, 2003 @09:06PM (#6576170)
    Cable modems generally ping better than DSL for whatever reason, and I'm sure even fatter dedicated lines are better as well.

    On my cable modem (adelphia) I get 10-12ms for the first 8 or so hops as they are all on the adelphia servers, after that I can get as low as 20ms or even 18ms for more local stuff (I get about 23 to www.yahoo.com). I live in San Diego and this type of service is only about $35/month. On my DSL (Pacbell) I used to get 15-20ms to the first hop even, whereas i get 9-11ms now.
    • Remember that the granularity (precision) of your "ping" times may be somewhat limited by the precision of your computer's clock. On many systems, these times are only accurate to about 10ms.

      And, of course, latencies change diurnally, i.e., over time, due to the changing traffic patterns throughout the day and week. Traffic levels through major ISP's will be higher during the day because of people surfing porn^H^H^H^H the web at work, but they may be higher for cable modem/DSL users during the evening, wh
      • WHAT??? your clock better be more accurate than ms's or your PC isn't going to be doing JACK. Linux IP stack is accurate to hundredths of a millisecond, although at that point you are measuring delays in packet processing in the NIC and kernal =) Trust me on our Gig-E network all the windows servers show 1ms to anywhere in the LAN, but Linux can tell me the real times and I can often tell how many switches I am going through =)
    • On my cable modem (adelphia) I get 10-12ms for the first 8 or so hops as they are all on the adelphia servers, after that I can get as low as 20ms

      Um... Wow?

      I also have cablemodem through Adelphia, and get FAR higher ping times...

      To just my segment gateway, I get 11ms. To the very first non-Adelphia node, I get 75ms. To a machine on COX's cablemodem network, I get as high as 120ms (speaking of which, back when I had COX myself, which used @home at the time, I got similar latencies).

      So I don't think

  • ISP's (Score:5, Informative)

    by BrookHarty ( 9119 ) on Wednesday July 30, 2003 @09:19PM (#6576252) Journal
    Verizon has 2 networks in our area, one is a T1 (fijitsu)based, the other is T3 (westell) based dsl modems.

    I was on the fitjistsu on the 768/128, about a 33ms ping to the seattle bbnplanet backbone, I moved down the street, and they put in the new higherspeed network. 1500/384 and 10ms to the bbnplanet backbone.

    USwest back in Spokane was about 15ms on a 768/768 cisco modem.

    While I find Verizon and other telcos to be better bandwidth and ping, smaller mom and pop ISP's tend to oversell. Speakeasy was would be choice if the telco is oversold, and earthlink if ISDN is your only choice. Thou small ISP's do re-sell ISDN cheaper, and ping is good enough for multiplayer games, 20ms+. (Remember its different for each user and location!)

    I'd check out dslreports [dslreports.com] and ask other people in your area. And networks change from city to city, cable/dsl/isdn/frame all depend on the routers and hop count. Plus if your ISP is a peering partner with local ISP's, they connect all major ISPs locally, thats a plus. Sometimes you notice crazy routing, like Seattle to California and back to go across town to an ISP without a local peering agreement.

    Also, you call your ISP, and ask them to do a traceroute from their network to a gameserver and email it to you. I've asked this from hosting services, and who they having peering agreements with. Some will even give you a network diagram or have them posted on the site, like Verio. (Who while expensive, does seem to have good peering agreements.)

    • Re:ISP's (Score:3, Informative)

      by cdrudge ( 68377 )
      Verizon has 2 networks in our area, one is a T1 (fijitsu)based, the other is T3 (westell) based dsl modems.

      The actual difference between the two is the Fijitsu is frame-relay based and the Westell is ATM.

  • Why VoIP an issue? (Score:4, Insightful)

    by 0x0d0a ( 568518 ) on Wednesday July 30, 2003 @09:36PM (#6576345) Journal
    VoIP shouldn't be an issue. An additional hundreth or fiftieth of a second is not noticeable.

  • count yourself lucky (Score:4, Informative)

    by elmegil ( 12001 ) on Wednesday July 30, 2003 @09:46PM (#6576399) Homepage Journal
    I had to fight and fight to get my ISP to take seriously my demand that the first hop be less than 50 ms or I was going to find someone else. See, I went with a provider that I thought was a local ISP who turned out to just be reselling service from...halfway across the country. So, I get the ATM link from here (Oak Park) to my CO, but I'm positive that that gateway router is in Virginia. If I wanted to give the business to my ILEC, I could probably do better, but as long as it's 50ms or less I can live with it. If I changed to Ameritech I'd probably have to give up my static IP and unblocked ability to have a small web server too.
    • Um, I work for ameritech (SBC actually). They offer static IPs. I'm not sure what kind of blocking they do on the residential accounts, but for business I highly doubt they would block ANYTHING, considering you are buying static IPs and those are typically only used if you want to run a server.
      • And what's the extra upcharge on that? I got my static at no extra charge whatsoever. And I am talking residential, not business.

        Just checked your DSL offerings, and something only roughly equivalent with no mention of static IP's is $10 more than what I pay today, and that's before all the myriad fees get tacked on.

  • Maybe when you're done bragging about your internet connection, you can come back down to earth. (i.e. the place where 12 ms on your first hop doesn't matter) What are you doing with this anyway? Playing games? 12 ms is not going to make any noticeable difference anyway. Get real.
  • Some ISP's who use satelite also offer a Gamers plan that doesn't use the satelite and has a much higher ping. Of course since your bypassing their bulk buying satelite power you either pay more money or/and get less bandwidth.

  • Latency and Throughput (Score:5, Informative)

    by blate ( 532322 ) on Wednesday July 30, 2003 @09:57PM (#6576460)
    Your post states that latency and throughput are unrelated. For TCP connections (FTP, HTTP, IMAP, POP, and many games), this is absolutely not true.

    The maximum possible throughput of a TCP connection is one "window" of data per round-trip time. The "window" size is essentially the amount of unacknowledged (ACK'ec) data that can be outstanding. This is often called the bandwidth-delay product, I think.

    What you need to take away from this is that even if you had infinite bandwidth between you and your peer, the throughput of a single TCP connection is upper-bounded by the delay product. For example, if your window size is 32KBytes (I'm going to use 32,000 to make the numbers prettier) and the round-trip time is 100ms, then you can transmit (or receive) at most 32KB * 10 = 320KB per second. To go faster, you have to either increase the window size (which consumes more memory) or decrease the round-trip time (which is sometimes impossible, since the speed of light is a constant, or so my physicist friends claim).

    A couple other points.

    You're probably not capable of noticing the difference between 10ms and 20ms in terms of response time for interactive applications, including online gaming. if it were 10ms vs 100ms or 200ms, then yes, but 10ms is less than one refresh interval on your monitor, so you really can't "see" the difference.

    As far as VoIP (IP telephony) and other multimedia network applications are concerned, again, you must consider the end-to-end latency (one-way delay) and/or the round-trip time, not the latency between you and some arbitrary router at your ISP.

    The phone companies spec their systems (or so I've heard) such that the *round trip* latency for a domestic call is always less than or equal to 100ms. We're talking POTS here, not cell service, which experiences higher latencies.

    I work on VoIP software; in an IP call (both ends are IP clients), it's very hard to keep the *one way* latency below about 100ms, if you're lucky, even if both clients are on a LAN. This is because you have to have various buffer and jitter compensation delays so that the sound quality is acceptible under somewhat adverse network conditions. In a typical call across the internet, 200ms one-way latency, IMHO, would be considered quite good.

    So your 20ms intra-ISP latency (vs. the 10ms that your friend reports) is in the noise.

    Oh, I should also mention, for completeness, that packet loss (or even reordering, which is more common that you may realize) can *really* hurt both TCP and VoIP (which usually uses UDP) performance/quality. This gets into some messier technical issues... basically, though, if your DSL isn't lossy, and you're getting 20ms intra-ISP latencies, you're doing as well or better than most of us.

    Your friends who are running on 56k modems, who eat 200ms just to get their packets to the ISP's router on the other side of the PSTN are really going to be hurting :)

    • Re:Latency and Throughput (Score:3, Insightful)

      by bwt ( 68845 )
      10ms is less than one refresh interval on your monitor, so you really can't "see" the difference.

      For gaming, though you often have human race conditions. The frame is drawn, the two players see each other for the first time and hit the "fire" button. Whoever gets the message to the server first kills the other. Taking a .01 second hit absolutely can make a difference even if it is less than the frame redraw time. Consider all the Olympic events and horse races etc... that have been decided by such margins
      • That's just bad game design. The packets should be timestamped, and the effect should correspond to the
        time of the cause, not the time of the arrival of the packet
        reporting the cause (within the limits of the jitter of the
        clock synchronization protocol, at least).
        • And thats insecure game design. I'd rather have a game boil down to whoever has lower latency than whoever can hack their client to mangle timestamps.

          A better method is what halflife does -- Check clients ping, check what they saw $ping ago, then process. The downside (for us LPBs, but upside for dialups) is I can run past a hallway, see an enemy a the other end, and keep running. then in ~400ms, I'm dragged back to the hallway because according to the server, they shot me.
    • This is often called the bandwidth-delay product, I think.

      Funny, I always thought it was called the window size.

      To go faster, you have to either increase the window size

      Or increase the MTU (bigger packets), which may not be a big problem if you're on DSL (short hop) connected to a fiber link (low error rate), set MTU discovery off, and clear the DF bit.

      • Or increase the MTU (bigger packets), which may not be a big problem if you're on DSL (short hop) connected to a fiber link (low error rate), set MTU discovery off, and clear the DF bit.

        Increasing the MTU will not necessarily improve throughput.

        If anything, a lower MTU could conceivably improve TCP throughput, since smaller packets get to the far end more quickly and would thus be acknoledged sooner, and much less total bandwidth is wasted on retransmissions of the (smaller) lost packets.

        Imagine a worst

    • Maybe *I* cant see the difference in 10ms and 20ms, but my computer can. If you play Battlefield 1942 on a LAN youll notice that the serverclient physics coordination allows you to stand on top of a moving vehicle, and the lower your ping is the faster the vehicle can be moving before you fall off.
    • Your post states that latency and throughput are unrelated. For TCP connections (FTP, HTTP, IMAP, POP, and many games), this is absolutely not true.

      Actually, in most digital communication systems, this is untrue, not just TCP/IP. Also on a hardware level, there is a trade off between latency and bandwidth - e.g. in microprocessor pipelines, bus interfaces. If it were possible to reduce latency and maintain the same bandwidth, they would do it.


    • Yes, latency and throughput are related, but
      The point about TCP windows is likely bogus for this application. Most modern TCP implementations include the window scaling option, which will allow scaling to quite high data transfer rates. At these low data rates (a few megabits, or even lower for games) the windows are unlikely to cramp your style (by limiting bandwidth) One usually wants bigger windows for high volume transfers (say > 10 Mbytes/second) that you would see on a LAN.

    • Something else to remember: If your DSL service has a bandwidth cap, your latency may shoot up once you exceed the cap. A DSL modem I had in college based the cap on the number of bytes sent/received per second. Once the threshold for the download cap was reached, it would not pass through any more downstream data for the remainder of the second.

  • It Depends (Score:3, Insightful)

    by suwain_2 ( 260792 ) on Wednesday July 30, 2003 @10:37PM (#6576636) Journal
    It depends on how big your ISP is. If they immediately feed you out onto someone else's network (ie, if they're a tiny ISP or whatnot), you'll get low pings (in theory). A larger ISP (Adelphia, in my case) has like 8 hops before I go onto above.net, averaging 39 ms until I'm off adelphiacom.net. Latency on your ISP's network isn't necessarily a meaningful measurement. I'd be much more interested in ping times to certain hosts. I average ~80 ms, although this can vary hugely -- if I'm pinging sites in Asia, it'll obviously be a bit bigger.

  • ADSL vs. SDSL (Score:5, Informative)

    by EinarH ( 583836 ) on Wednesday July 30, 2003 @10:41PM (#6576654) Journal
    Have you considered changing your ADSl into either a SDSL or VDSL service?
    Some of those ISP's that offer ADSL have started to offer SDSL or VDSL. VDSL is currently very expensive in my area and only people within a short distcance from a telephone central can get it. SDSL is more flexible when it comes to max distace. Most people on SDSL get lower ping.

    When I got my new connection I could either choose between 1024/512 ADSL at $85 or 1024/1024 at $140.
    A bit expensive, but I get my own permanent IP, no pay per GB thing, can have my own servers etc.
    And I can't complain at the latency, since many of the other users on the ISP are offices and bussiness whom almost only use their computers at office hours I get very low latency. Approx. 15 ms. to many CS-servers and the same to a backbone.

    So I'm happy, but I still gaze at the connection of a friend of mine. He just got a VDSL 12500/6250 at $227. Officially, According to their User Agreement he cant't resell but the ISP is not that strict on it so he allready has 10+ customers... ;-)

    • He just got a VDSL 12500/6250 at $227.

      If you don't mind my asking, where does your friend live (in general -- country and city), and what kind of hurdles did he have to jump through to get it?

      Yes, I'm jealous too. :)
      • Ahh, I forgot the whole thing.
        Sorry for late reply.

        He lives in Norway and his new ISP is Firstmile. (www.firstmile.no). Its a new ISP, a subsidiary of ZyXEL Comunnications. Unfortunately, they only deliver in limited parts of Norway.

        Products and pricing at the bottom of this [firstmile.no] page. Prices in NOK, so you have to divide with 7 to get $.

        No special hurdles, but he had to terminate his old ISP-agreement and pay them 3-months extra because of a 2 year agreement setup.

  • 22ms is decent indeed and will serve you more than well for FPS games and VoIP. For ATM networks, the maximum latency for voice is defined as 500ms. People can get by with 200ms and not know it unless theyre playing reflex games on the phone.

    No its not about time we get focused here, when ISPs were over 500ms it was an issue. Below 50ms theres no issue at all unless you just WANT to have lower latency for the sake of it. And then counting hops and demarcating the bounds of your ISP gets you nowhere. If Sym

  • Latency to where? (Score:5, Interesting)

    by cperciva ( 102828 ) on Wednesday July 30, 2003 @11:51PM (#6577063) Homepage
    Latency-to-edge-of-network has got to be the most broken benchmark I've ever seen. If your network passes its traffic off to a different network within the same city, while my network takes it halfway around the world and passes it directly to the destination machine's network, my packets are going to be staying within my network for a long time... but they'll probably reach their destination sooner than yours.

    If you're going to measure how long it takes for your packets to get somewhere, make sure you also measure where your packets are getting to.
  • I remember a couple years ago reading a PC Gamer QA/Article on Broadband types, they mentioned the whole DSL latency thing as if it were inherent. This to me makes the lower cost somewhat irrelevant when my cable network infrastructure generally leaves me with real world pings of under 100ms.

    The added latency in dsl setups can also add to irritation when web surfing as well.

  • You shouldn't have too much of a noticeable problem with voip with a 22ms latency. Now if the latency fluctated between 10~100 ms sporadically, you'd probably have intelligiblity issues...
  • Your ISP may be better connected to the backbone
    than your friend's ISP. What matters is not the latency
    to the AS boundary, but the average latency to your
    peers. Also, 8MB up with a 128k down is not going to
    get much better ping time than 512k up with a 128k
    down: The 128k down segment is going to dominate
    your ping time (which is bidirectional).

    Yes, latency sucks. It's sucking more and more as
    ISPs optimize devices for b/w at the expense of latency,
    although the customer base would benefit more from
    decrease
  • Is there an easy way to run a pathping in linux? I suppose I could run traceroutes and pings manually or with a script and try to reconstruct what pathping does (according to the M$ site).
    It would be nice to have a way to do it because pathping seems as useful a utility for network admin as nmap, etherape, and ethereal.

    • Free equivalent (Score:3, Informative)

      by 0x0d0a ( 568518 )
      You're looking for the excellent mtr [bitwizard.nl].

      Believe me, there isn't anything you can do on a network in Windows that you can't do better in Linux.
  • 22 ms is pretty good. However, as everything, it depends. For example, how large is your ISPs network and how close does it get you to the final location you are interested in? For example my cable ISP has a larger network. If I try to contact a server a few states away, it uses my ISPs lines for most of the trip.

    If you have a service level agreement, it usually specifies 100ms as maximum round trip time within the ISPs network. I guess they pick this rather high number as it usually is fast enough and sho

  • 8mb dsl latency (Score:2, Informative)

    by BTF ( 694018 )
    22ms is definitely good. Especially leaving your isp.

    Dsl speeds won't affect latency too much. I know it's not supposed to, but it does.
    A 1184/160 dsl will ping around 15ms to its gateway
    A 1728/384 dsl will ping around 11ms to its gateway
    A 3488/800 dsl will ping around 7ms to its gateway

    I have a feeling its more related to the upstream speeds than the downstream. An 8MB dsl has an 800 upstream maximum so the pings will most likely be the same as a 3MB dsl. Isp's can have different upstream spe

  • router equipment (Score:2, Informative)

    by blosphere ( 614452 )
    Your 22ms is not that bad... but you can get it down to 10ms with fast-path. I was able to squeeze out to 7-9ms (on first hop), but then I have heavily tweaked my own dslam profile (used to work for an dsl provider).

    The other thing is, that you shold really only be interested in end-to-end RTT, not the individual hops. For example, if there's cisco 4xxx series switches with SUP-3's out there, your icmp/traceroute IP packets gets processed in the processor card, not on the interface, causing an 10ms more l

  • I have speakeasy sdsl [shameless plug] and have a 10ms time to google.

    For games, it's ~50ms for anything on the west coast, and ~10-30ms for anything on speakeasy's network.
  • I just did a quick traceroute from the webserver here where I work through a 1.5Mmbs or so ADSL w/ Qwest (up and downstream pretty much the same) to my home computer (standard Comcast Cable). To my suprise, from the router here to its first hop was +38ms... but from my home connections gateway to my home computer is only +10ms. Yes, I did it a few times just to make sure it wasn't a fluke... but it was pretty consistent around that.

    This is a very expensive ADSL line that we have going here... I would think


  • I heard some MCI execs were hoping for a latency of 5 years + good behavior.

  • Trade? (Score:3, Funny)

    by KillerHamster ( 645942 ) on Thursday July 31, 2003 @04:49PM (#6582977) Homepage
    If you aren't happy with your DSL connection, I have a very nice 14.4 modem that I would be happy to trade you.
  • What would be a good tool for me to use in 'nix to figure out the basic latency of my connection? I suppose I could just ping out a well-known host, but that would also involve the latency at their end?
  • % sudo mtr www.yahoo.com

    Packets Pings
    Hostname %Loss Rcv Snt Last Best Avg Worst
    1. myhost 0% 9 9 0 0 0 2
    2. sfo1.dsl.speakeasy.net 0% 9 9 14 13 14 15
    3. border5.g3-4.speakeasy-29.sfo.pnap. 0% 9 9 14 13 18 46
    4. core4.ge2-0-bbnet2.sfo.pnap.net 0% 9 9 15 14 51 207
    5. so-1-3-0.0.ar4.sfo1.gblx.net 0% 9 9 17 14 21 62
    6. pos1-1.core1.SanFrancisco1.Level3.n 0% 9 9 14 14 16 18
    7. so-4-0-0.mp2.SanFrancisco1.Level3.n
  • The first thing that makes a VoIP call bothersome as latency rises is the echo. If the person you talk to has a good echo canceller, you will be OK up to about 150msec. After that, you start to attribute delays in each others' reactions to emotions, usually reluctance, reulting in anger.
  • Everyone here seems to be suffering from the same delusion that a fatter pipe means a faster line.

    Boy have a I got bridge to sell you! ( and some terrific coastline property too!)

    No matter how fast you can download data, all the signaling involved has to obey the laws of physics.

    Expecting to get less than 20ms of latency getting from your pc to your ISP's connection to the NET is extremely unrealistic (the average T1 introduces 20ms of latency) just because of the number of signal + data processing devic
  • From an engineering point of view, it depends upon your requirements, but for public access internet providers, lowish latency is important.

    We use the term "bandwidth delay product" rather than "bandwidth" as this refects the combination of speed and latency.

  • traceroute to slashdot.org (66.35.250.150), 30 hops max, 38 byte packets
    1 my.gateway (216.xxx.xxx.xxx) 0.690 ms 19.335 ms 0.404 ms
    2 some.machine.at.savvis.net (216.xxx.xxx.xxx) 1.929 ms 1.903
    ms 2.108 ms
    3 500.POS2-1.GW4.ATL3.alter.net (157.130.81.41) 2.266 ms 2.175 ms 2.007 ms
    4 147.at-1-0-0.XL3.ATL1.ALTER.NET (152.63.81.50) 2.449 ms 2.797 ms 2.680 ms 5 0.so-7-0-0.XL3.ATL5.ALTER.NET (152.63.85.190) 3.104 ms 3.372 ms 3.207 ms
    6 193.ATM6-0.BR1.ATL5.ALTER.NET (152.63.80.113) 3.139
  • Are you kidding? 22ms is great for voice chat. Think about it ... You're talking about 1/50th of a second here. You really can't tell, trust me.

    The main culprit in VoIP latency is really jitter. That's basically packets that arrive out of order or don't get there and need to be resent. If you've got a lot of jitter it drives up the latency to compensate (the codec needs time to reassemble them in the right order). It's better often just to drop missed packets ... there's lots of work in this area.

    Jitter i
  • While some may call this nitpicky, I constantly am on the receiving end of many many otherwise informed folks that think that ICMP PING is an accurate test of network performance.

    It's not.

    PING was intended for reachability checking, and as a secondary feature, response time.

    The ICMP part of most IP stacks often has the lowest priority to receive CPU time in a lot of IP stack implementations. When you PING, the return packet back to you is at the mercy of the resources of the system writing, generating, a

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