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mikebrown Posted:
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Haniltery Posted:
For wipe call
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usually apply to
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dconnor Posted:
What is the main
number on the
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Trafford Posted:
Seems like a
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rely exclusively
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diazou Posted:
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On Mar 28, 2017 at 12:42:33

jeddaisg Posted:
Hi all We have
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Vonage Great job Scerruti!! Here is the information I culled from TestMyVoIP.com



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



Read The Full Thread:

How to Interpret Speed Test Results


The speed test hosted on this site is an applet named "MySpeed". More information
In general it is hard to define "acceptable" results for the parameters for some
Forget the phone, try Pony Express Like tin cans and string As bad as a crummy

JoeP posted "Great job Scerruti!! Here is the information I culled from TestMyVoIP.com" on 09/19/2005

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