Maximum Latency for ISPs?

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?"

Comments

"Normal" 8Mb?

[leonbrooks]

I've got 512/128kb and consider it to be luxury. Perth, West Oz.

Well...

[Anonymous Coward]

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?

CAP or DMT?

[austad]

What transmission scheme are they using? With CAP, you can expect lower latencies, around 10ms I think. However, most telcos are switching to DMT because I think it's more scalable. Unfortunately, DMT gives crappy latency, I've seen 60ms in some cases.

22ms latency to leave your ISP's backbone is actually quite good for DSL.

Featurewise, most cable modems are crappy, but their latency is better than DSL in most cases.

As far as VoIP goes, 300ms will still give good results. Some codecs don't play nicely with high latencies, but I've used VoIP with a 600ms latency satellite link, and it worked just fine. The latency on your TDMA or GSM phone is several hundred ms, just call another cell phone from yours and put one up to each ear and talk, there's almost a second delay.

Boohoo..

[OutRigged]

You've got an 8mbit a second ADSL connection, and you get 22ms pings? Cry me a river.

Alright, yeah. I'm jealous. :(

22ms?

[alph0ns3]

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!

ISP's

[BrookHarty]

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 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.)

Latency and Throughput

[blate]

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 :)

ADSL vs. SDSL

[EinarH]

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... ;-)

Latency to where?

[cperciva]

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.