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Julianp
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Posts: 27

PostPosted: Fri Jun 09, 2006 12:14 pm    Post subject: Reply with quote Back to top

MrMark wrote:
Well...some might say I'm nuts, but even so, I don't think I am crazy enough to do 24/7 testing using a manual test. It would need to be automated.

I already own PingPlotter, if that will do the job. I would just need to know what to ping, and what parameters to use.


No, as far as I am aware Ping Plotter won't do QoS testing but will do latency. Try MYVoIPSpeed Advanced PC product, it is free for 2 weeks usage which is ample time and it provides a managed testing option. http://www.myvoipspeed.com/pc/index.html
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scerruti
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PostPosted: Fri Jun 09, 2006 3:13 pm    Post subject: Reply with quote Back to top

The ping plotter settings I have recommended in the past (based on a forum search are:

Quote:
You will want to configure it to use UDP packets (Advanced Options -> Packet Options -> Packet Type) and try using 64.210.19.18 as your destination address.


While julianp is absolutely correct that PingPlotter is not going to give you a qos measurement, it is going to give you latency and dropped packet measurements and, more importantly in my opinion, the location where the problem exists. In addition, with PingPlotter you have the flexibility to test towards a Vonage endpoint (as above) or towards the speed test endpoint. JulianP, how is the test destination specified for MyVoIPSpeed?

Additionally I don't think PingPlotter will model Voip traffic as well as the MyVoIPSpeed test will. The answer may in fact rely on alternately running both of them to get a clear picture.

If you actually have a problem where packets didn't get dropped and latency wasn't changing significantly but you had significant amounts of jitter I would be surprised if you could get an ISP to correct the issue anyway.

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MrMark
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PostPosted: Fri Jun 09, 2006 3:32 pm    Post subject: Reply with quote Back to top

Maybe my question should be, if I run PingPlotter for a day (or a week), who is going to analyze the results?

Even though I have a significant background in I.T., I don't think think I am qualified to interpret the results. I mean, I can see if a node it particularly slow, but once I see that, I'm not sure what I would do after that.

I could also go for the MyVoIPSpeed PC application, but I am guessing that would leave me with the same dilemma...how to interpret the results, and what to do about it.
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MrMark
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PostPosted: Fri Jun 09, 2006 6:13 pm    Post subject: Reply with quote Back to top

FYI: When I run PingPlotter against 64.210.19.18, the packet losses at the destination address are massive - around 68 percent.

I did this during a Vonage call, and the call quality sounded excellent, so I'm going to have to say that pinging to that address does not relate to voice quality.
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scerruti
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PostPosted: Fri Jun 09, 2006 7:11 pm    Post subject: Reply with quote Back to top

Yeah, the last hop doesn't always work, you can choose the hop before that as the destination.

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Julianp
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PostPosted: Fri Jun 09, 2006 7:35 pm    Post subject: Reply with quote Back to top

MrMark wrote:
FYI: When I run PingPlotter against 64.210.19.18, the packet losses at the destination address are massive - around 68 percent.

I did this during a Vonage call, and the call quality sounded excellent, so I'm going to have to say that pinging to that address does not relate to voice quality.


Correct. It has no bearing except to measure the RTT which is important. Also 68% is not a firewall issue that would record 100%. 68% indicates packets are dropping big time, very unhealthy, but 32% are getting through.

The fact that you lose packets does not detract from understanding your RTT. ICMP traffic being diagnostic traffic is not afforded the same privilige as UDP or TCP traffic. In simple terms UDP and TCP packets usually get to use the HOV lane and make it to the destination. ICMP on the other hand does not when traffic is high and in some cases will fail to reach the destination. Packet loss is very bad and if left uncorrected will eventually affect the Voip. YOU SHOULD NOT DROP PACKETS except by firewall design. It is an early warning, first comes high and erratic RTT then eventually packet loss.

The speed test does not use ICMP (Ping) for this reason it uses UDP and TCP. That said ICMP is an excellent early warning protocol because it is the first packet type (normally) to be victimized. When you get to the stage that UDP and TCP packets start dropping you should packup and go home.

Knowing your normal RTT using PING does provide a valuable and useful baseline measurement that alerts you to when there is trouble coming. Knowing the values that are normal allows you to understand when the values are abnormal. More importantly it also allows you to validate 'normal' as 'acceptable' because it often isn't.
To make my point say you own a BMW 530i and have done so for 6 months. I think it would be fair to say that you would have a good feel for its normal gas consumption. Just say in the 7th month you suddenly notice that your gas consumption drops from the normal 30 MPG to 10 MPG, wouldn't you take it to the dealer and say there is someing wrong fix it. Maybe if you purchased the BMW and it had only 10 MPG you would have accepted it as normal but it wouldn't necessarily be acceptable. The RTT is a measure similar to the MPG of the car, knowing what it is (average) and what is acceptable is an important baseline measurement.

When all is said and done what matters is the jitter, it is singularly the most important metric for Voip. Jitter is in essence the variation between what is normal and what is abnormal. i.e it is the variance above normal. If the variance is high then Voip will suffer. To know high you must know normal.

You cannot allow 68% packet loss to continue it will affect you if left uncorrected. Ultimately the ISP responsible must accept responsibility.
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Darrell_G
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PostPosted: Sat Jun 10, 2006 8:17 am    Post subject: Reply with quote Back to top

I thought I would post my results here...they seem to be in line with the results I get from testyourvoip.com.

Speed test statistics
---------------------
Download speed: 8397904 bps
Upload speed: 739344 bps
Quality of service: 96 %
Download test type: socket
Upload test type: socket
Maximum download pause: 180 ms
Average download pause: 64 ms
Minimum round trip time to server: 43 ms
Average round trip time to server: 49 ms

Voip test statistics
--------------------
Jitter you --> server: 5.7 ms
Jitter server --> you: 9.1 ms
Packet loss you --> server: 0.0 %
Packet loss server --> you: 0.0 %
Packet discards: 0.0 %
Packets out of order: 0.0 %
Number of supported Voip lines: 9
Estimated MOS score: 4.0
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MrMark
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PostPosted: Sat Jun 10, 2006 10:49 am    Post subject: Reply with quote Back to top

For what it's worth, I have PingPlotter configured to use UDP packets.

Following the car metaphor, when driving to 64.210.19.18, it is, for the most part, a smooth ride all the way to the destination. There is no packet loss until my trace tries to turn into the driveway at 64.210.19.18.

Here is a link to a screen shot:

PingPlotter Trace to 64.210.19.18


That leads me to believe that there is too much traffic at that destination. The fact that the trace results do not relate to actual Voip results leads me to believe that this is either not the same destination that my voice packets are using, or that the voice traffic is somehow treated differently than my UDP trace traffic. In either case, it makes the test kind of useless for evaluating Voip.

By comparison, when I trace to 205.234.111.151, the round-trip is relatively uneventful:

PingPlotter Trace to 205.234.111

The average round-trip time is much lower at 59 ms, and the "jitter", as indicated by the black graph line, is obviously much lower. There appears to be some packet loss at the border between savvis and defenderhosting, so all of the packet loss beyond that point has to be taken with a grain of salt.

The packet loss at the destination is considerably lower than at the border, indicating that the traffic is moving past the border...so it seems that some of the defenderhosting routers are simply not responding to all of the pings.
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Julianp
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PostPosted: Sat Jun 10, 2006 12:14 pm    Post subject: Reply with quote Back to top

MrMark wrote:
For what it's worth, I have PingPlotter configured to use UDP packets.

Following the car metaphor, when driving to 64.210.19.18, it is, for the most part, a smooth ride all the way to the destination. There is no packet loss until my trace tries to turn into the driveway at 64.210.19.18.

Here is a link to a screen shot:

PingPlotter Trace to 64.210.19.18


That leads me to believe that there is too much traffic at that destination. The fact that the trace results do not relate to actual Voip results leads me to believe that this is either not the same destination that my voice packets are using, or that the voice traffic is somehow treated differently than my UDP trace traffic. In either case, it makes the test kind of useless for evaluating Voip.

By comparison, when I trace to 205.234.111.151, the round-trip is relatively uneventful:

PingPlotter Trace to 205.234.111

The average round-trip time is much lower at 59 ms, and the "jitter", as indicated by the black graph line, is obviously much lower. There appears to be some packet loss at the border between savvis and defenderhosting, so all of the packet loss beyond that point has to be taken with a grain of salt.

The packet loss at the destination is considerably lower than at the border, indicating that the traffic is moving past the border...so it seems that some of the defenderhosting routers are simply not responding to all of the pings.


As you point out it is not possible to say that the packets were lost on the last hop, what you can say is that the packets were lost on their way to the last hop. Because they are lost you never know where the loss takes place you just know the target destination when the loss was reported. Although as the value is high and other hops are low this definitly reinfoces the point. In argumant to that it is also true to say last hop destined packets have the highest exposure for loss because the journey is at its longest point. Secondly a UDP trace and indeed a TCP trace is likely to have issues with the very last hop because it is the last hop and packets never pass through to another hop further along. The way the trace works is that the TTL is set to expire one hop further than the previous hop. This causes an ICMP(note not UDP but ICMP) expire in transit message to return. This means that the trace is is operating is two modes 1. where UDP packets are passing through on their way to another detination (using the HOV lane) and one where this router is the destination and the TTL has expired. In a situation where a UDP packet expires at a hop the transaction is degraded because the router now knows that this packet has failed (no more priority for this packet). I would therefore hazard a guess that there is a considerable delay at the Vonage end of things in hadling failed UDP packets this delay may well exceed the timout of the pingplotter trace hence the packets goe missing.

It would be interesting to try this. 1. Set ping plotter to ICMP to see if the profile of the two traces change. 2 set a simple DOS ping to that destination also :>ping -n 20 64.210.19.18 to compare

Another small point of note is that Pingplotter palces 0-200ms as acceptable, for me it is not, anything over 110 I consider bad, prefereably <100ms is acceptable. Some of your high latenies are very high indeed. Clearly your ISP has issues with 10.53.160.1, thi sis the first hop and it shows unacceptably high peaks appraching 200ms, also 4.71.12.17 is consistently unnaceptable > 300ms on both.

If you take a look at http://205.234.111.216/data.html you will see our measure consistency. The plot high blocks are when the automated Voip test runs (every 5 minutes). Below you will see an intersting time correlation provided by those automated Voip tests, My time-zone is GMT but the data is EST so if you add 5 hours and look at the download speeds and QoS you will notice the impact between the 9-5 business day. by 10am the line degrades < .5mbit by 6pm it is back to normal 4mbit. A clear definition of the baseline and clearly an ISP iproblem tp be solved.
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MrMark
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PostPosted: Sun Jun 11, 2006 11:05 am    Post subject: Reply with quote Back to top

It’s too bad that I/we don’t have access to the perfect test, but right now, PingPlotter seems to be giving me the best picture of any of the testing options that I have available to me. The MySpeed test may use internal information or criteria that PingPlotter does not, but since that information is not presented to me, all I can tell is that my connection tests badly. I really need a tool that will lead me to a solution, rather than just reporting that there is an unknown problem.

Whatever the cause of the packet loss, tracing to 64.210.19.18 is problematic if only because of the fact that there are so many packets being lost that it makes the graph hard to read! Is there no better destination to trace to?

If I set PingPlotter to use ICMP packets, that seems to eliminate the packet loss problem at the destination, but if I am trying to test for UDP traffic, I’m not sure what that proves. Pinging from a command line returns similar results.

The default “acceptable delay” is 200 ms, but it can be configured to show whatever one wants it to show. My average round-trip on the 205.234.111 server trace was only 59 ms whereas the trace to the Vonage server averaged 103 ms. It is true that there were some delays on the first hop but I don’t think it’s fair to say that this indicates an ISP problem any more than it would be fair to say that the Vonage server is dropping 65 percent of the packets. The fact that successive hops do not show any significant errors or delays indicates to me that the router simply does not always respond to the pings in as timely a manner as I would hope. Admittedly, there is no way to say for sure with the server dropping so many packets, but it seems to me that most of the delay and errors occurred at the final destination, since that is the boundary where the average round trip time gained 37 ms. That is one reason that it would be much better to trace to a server that is more responsive.

I don’t know what 4.71.12.17 is, but apparently it’s part of Level3.net. My interpretation of the results is that most of the delay in this trace (other than the Vonage server) is within the level3 network. I’m not sure if that’s Roadrunner’s fault or not and I would not know who to call to gripe about it.

I followed your link, but I am not familiar with that software, and I have to admit that I did not understand exactly what I was looking at. My only comment would be that although I can certainly see that the performance suffers at certain times, I cannot see the source of the problem…you may have access to more information that is not presented in the linked page, but I, personally, would not be able to say that the ISP is the source of the problem by looking at this information.
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