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scerruti
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Joined: Feb 05, 2005
Posts: 1424
Location: Carlsbad, CA (finally)

PostPosted: Mon Sep 19, 2005 9:24 am    Post subject: How to Interpret Speed Test Results Reply with quote Back to top

JoeP wrote:
I am looking for an accurate interpretation of the results of this speed test and the TestMyVoIP.com test. What's good, bad, acceptable. Also, when a particular value is determined to be problematic where can it be found as to the cause and recommended solution.

I was not able to find this information anywhere to date. It would seem to me that it would be a "sticky" for newbies like me and to avoid having someone ask the same question over and over.

Thanks

Joe P

Download 691,145 to 3,017,424 bps
Upload 370.467 to 475,168 bps
QOS 18 to 91%
RTT 56 to 60 ms
MaxPause 71 to 190 ms

Note: As you can see my test results are all over the board and I am trying to accurately evaluate and understand them.


The speed test hosted on this site is an applet named "MySpeed". More information about MySpeed can be found at www.myspeed.com.

This test measures the available bandwidth between your computer and the web server hosting the test. The test results can be affected by a variety of factors including other applications running on your local machine, congestion at your ISP, specific bandwidth limitations between you and the target site and the load on the hosting server.

I am going to provide a very brief description of the test outputs here. For full descriptions please look at the FAQ.

  • Download: speed of an incoming http transfer
  • Upload: speed of an outgoing http transfer
  • QOS: quality of service is the minimum speed seen during the test divided by the maximum speed
  • RTT: round trip time, the total time it takes from sending a packet to receiving a response
  • MaxPause: longest single delay during transmission measured when reading data from the incoming sockets


The importance of these measurements? Obviously a certain minimum bandwidth in both directions is required for a Voip call. More importantly you need a high QoS number. A low QoS number will result in choppy, and possibly dropped, calls. When the QoS number is low packets are being explicitly dropped, or delivered out of order in which case they are also dropped. You could have a consistent good quality connection with a high RTT (this results in latency). On a Voip call latency expresses itself as a delay between the speaker and the listener. This phenomenon is also present in PSTN calls, particularly overseas calls, because of long transmission lengths or satellite relays. MaxPause is essentially a brief interruption of your connection. This can be caused by network congestion or by other applications running on your local machine. On Voip calls a large MaxPause results in drop outs for the length of the pause, and if long enough, a dropped call.

Acceptable Values

Vonage wants a minimum of 90 kbps in each direction to ensure high quality calls.

QoS: Should be in at least the 80's. However people have reported success with much lower scores. Unfortunately this test measurement is not useful unless you determine what is causing the low score. I don't put much stock in this QoS mechanism because it can be affected by many different causes and it is not based on a big enough sample to accurately reflect line quality.

RTT: Most callers notice round-trip delays when they exceed 250mSec, so the one-way latency budget would typically be 150 mSec. 150 mSec is also specified in ITU-T G.114 recommendation as the maximum desired one-way latency to achieve high-quality voice. *

MaxPause: Would ideally be less than 100. Meaning no dropouts more than 1/10th of a second. (Guess, refinements anyone?)

Action to take if you get inconsistent results from the Speed Test

  1. Make sure that no other applications are running on your machine and repeat the speed test. This includes network applications such as email programs and IM clients, but also non-network programs like word processors and spreadsheets.
  2. Most network problems occur on the link from your house to the ISP, particularly in the case of cable customers. File a complaint with your ISP and get them to look at the problem.
  3. You can use a more detailed test such as PingPlotter to tray to find out exactly where and when you are having problems.


Why run the test at testyourvoip.com

This test more closely mimics the network activity experienced during a call. It will provide more explicit details about exactly what the problems are that are affecting your Voip call quality, but it does not identify the source of the problem.

Ping Plotter

By configuring PingPlotter to run using UDP packets to a network point near the Vonage system you can identify the location of a network problem. Because PingPlotter can be run continuously it can also check for problems that are periodic or intermittent.

Other notes

Run speed tests at multiple sites to eliminate problems at the desitnation site or on parts of the network in between. For example, something at the edge of Comcast's network is capping my upload speed at 2 Mbps to the vonage-forum site, but not to the Vonage site. Note that this also affects my QOS score detrimentally, this is another reason I don't place much emphasis on the applet QOS score, my Voip calls would never push that much data and so would never be capped.

Run speed tests that mimic the actual traffic that you expect. Test to the Boston server at TestYourVoip.com because it is close geographically to Vonage in New Jersey.

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JoeP
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Joined: Sep 14, 2005
Posts: 22
Location: West Palm Beach, FL

PostPosted: Mon Sep 19, 2005 11:29 am    Post subject: Reply with quote Back to top

Great job Scerruti!!

Here is the information I culled from TestMyVoIP.com but I was unable to get the Acceptable Values:


TestMyVoIP.com Test Results Defined


From You to the Testing Location:

Post-Dial Delay is the time it takes, after you've dialed, for the phone you're calling to ring.

Call Setup Delay is the full time it takes to setup the call and receive an acknowledgement from the far end that it has accepted the call. Call setup delay includes post-dial delay.

Media Delay includes the full call setup time plus the time it takes to receive the first packet of media (conversation). Media Delay includes both call setup delay and post-dial delay.



From the Testing Location to You:

Post-Pickup Delay is the time that elapses between answering the call and receiving the first packet of media (conversation).

Call Setup Delay is the time from receiving the request for the call until the final acknowledgment from the caller that the call setup has been successfully completed.

Media Delay is the time between receiving the initial call notification request to receiving the first media packet (conversation). Media Delay includes both the call setup delay and post-pickup delay.

Note, that call setup and media delay measurements incorporate the effects of network impairments and signaling infrastructure (that is, the delays you care about) but do not reflect any delay in answering the call that might be injected by a slow user.

****************************************************************************

Here is the information I have on latency, Jitter and Packet Loss. It is rather lengthy and probably needs to be edited.


Latency, Jitter and Packet Loss

Once a call has successfully been setup, latency, jitter, and packet loss effects are important predictors of overall call quality

Latency: A measure of the delay in a call. We measure both the round-trip delay between when information leaves point A and when a response is returned from point B, and the one-way delay between when something was spoken and when it was heard. The largest contributor to latency is caused by network transmission delay. Round-trip latency affects dynamics of conversation and is used in our MOS calculations. One-way latency is used for diagnosing network problems.With round trip latencies above 300 msec or so, users may experience annoying talk-over effects.

Jitter: Jitter refers to how variable latency is in a network. High jitter, greater than approximately 50 msec, can result in both increased latency and packet loss. Let's see how.When talking to someone it's important that they hear what you say in the same order that you say it, otherwise they won't understand what you're telling them. Unfortunately, jitter causes packets to arrive at their destination with different timing and possibly in a different order than they were sent (spoken), with some arriving faster and some slower than they should.To correct the effects of jitter, Voip endpoints collect packets in a buffer and put them back together in the proper timing and order before the receiver hears them. This works, but it's a balancing act. Processing that buffer adds delay to the call, so the bigger the buffer, the longer the delay. Remember the effects of latency? Keep in mind, no matter how big the buffer is, it is finite in size. If voice packets arrive when the buffer is full then packets are dropped and the receiver will never hear them. These are called discarded packets.

Packet Loss: Just as it's important to hear what someone says in the order they say it, it's also important to hear all of what they're saying. If you miss one out of every 10 words or 10 words all at once, chances are you're not going to understand much of the conversation. This is packet loss — some of the voice packets are dropped by network routers or switches that become congested (lost packets), or discarded by the jitter buffer (discarded packets).Knowing the average packet loss for a call gives you an overall sense for the quality of the call. A call with less than 1 percent average packet loss will always sound better than a call with 10 percent loss. But average loss doesn't tell the whole story. You need to know what type of packet loss you encountered.There are two kinds of packet loss: "random" and "bursty". Think about two calls each with average 1 percent packet loss. Call A loses one in every 100 packets over the entire call (random loss) while Call B loses 100 packets in two clumps at the beginning and the end of the call (bursty loss). Which call would you rather have? That's why we report not just the average packet loss but also the type of loss and information on any bursts of packet loss during your call (reported as loss periods). It matters.

************************************************************************************

Based on the information you posted and the information I gathered it appears that I have an issue with outgoing calls. I will post in another thread and leave this one solely for the purpose of defining the results of the Voip tests.

Joe P
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scerruti
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Joined: Feb 05, 2005
Posts: 1424
Location: Carlsbad, CA (finally)

PostPosted: Mon Sep 19, 2005 1:59 pm    Post subject: Reply with quote Back to top

In general it is hard to define "acceptable" results for the parameters for some of these tests. In some cases it is because the test results do not directly relate to Voip performance, in others it is because the parameter may indicate a level of call quality that is acceptable to some, but not others.

That is why testyourvoip.com displays your score as it compares to fuzzy benchmarks.

Quote:
  • Forget the phone, try Pony Express
  • Like tin cans and string
  • As bad as a crummy cell phone call
  • As good as a decent cell phone call
  • Like calling next door
  • Better than being there!

_________________
Stephen P. Cerruti (ISP: TWC)
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JoeP
Vonage Forum Associate
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Joined: Sep 14, 2005
Posts: 22
Location: West Palm Beach, FL

PostPosted: Mon Sep 19, 2005 2:45 pm    Post subject: Reply with quote Back to top

scerruti wrote:
In general it is hard to define "acceptable" results for the parameters for some of these tests. In some cases it is because the test results do not directly relate to Voip performance, in others it is because the parameter may indicate a level of call quality that is acceptable to some, but not others.

That is why testyourvoip.com displays your score as it compares to fuzzy benchmarks.

Quote:
  • Forget the phone, try Pony Express
  • Like tin cans and string
  • As bad as a crummy cell phone call
  • As good as a decent cell phone call
  • Like calling next door
  • Better than being there!


What about the "See Detailed Results, (see my last post "TestMyVoIP.com Test Results Defined") such as Confused??

MOS Analysis from You TO Boston

Media Quality MOS 4.1 / 5.0
(Best with G.711 is 4.4)



Degradation Sources
Codec 0.58 62.6%
Latency 0.00 0.0%
Packet Discards 0.34 37.4%
Packet Loss 0.00 0.0%

Codec G.711 (PCM at 64kbps,
20ms RTP payload,
80kbps IP BW)
Round-Trip
Latency 280 ms
Packet Discards 1.2%
Packet Loss 0.0%
Loss Periods Min: 20 ms
Avg: 20 ms
Max: 60 ms
Random Loss

Jitter Min: 0 ms
Avg: 4 ms
Max: 60 ms

Signaling Quality
Post-Dial Delay 110 ms
Call Setup Time 120 ms
Media Delay 320 ms

MOS Analysis FROM Boston To You

Media Quality MOS 4.3 / 5.0
(Best with G.711 is 4.4)



Degradation Sources
Codec 0.57 79.5%
Latency 0.00 0.0%
Packet Discards 0.15 20.5%
Packet Loss 0.00 0.0%

Codec G.711 (PCM at 64kbps,
20ms RTP payload,
80kbps IP BW)
Round-Trip
Latency 180 ms
Packet Discards 0.9%
Packet Loss 0.0%
Loss Periods Min: 20 ms
Avg: 20 ms
Max: 40 ms
Random Loss

Jitter Min: 0 ms
Avg: 4 ms
Max: 40 ms

Signaling Quality
Post-Pickup Delay 162 ms
Call Setup Time 168 ms
Media Delay 195 ms
********************************************************************************

In those results above I am using the line at the time. I have saved other tests when the line is idle. Both tests show a significantly high RTT, Outgoing Packet Discards, and Media Delay. I also get complaints from those I converse with that they lose some of my speech, "it clicks off", "the volume drops", etc.

While I was conversing I ran 8 Vonage Speed Tests and the RTT ranged from 57 to 60, QoS 16-28, and the Max Pause 71-551 (4 of the 8 were in excess of 500ms), Upload ranged from 381-401 and the Download was 1.2 to 2.6.

Once the line went idle I ran two Vonage speed tests and got the following:

Download 594 bps
Upload 458,306 bps
QOS 59%
RTT 55 ms
MaxPause 621ms

Download 2,505,872 bps
Upload 456,336 bps
QOS 20%
RTT 58 ms
MaxPause 100 ms





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