Frequently Asked Questions

So, how does it work?
Glad you asked. TestYourVoIP.com is supported by a VoIP performance management system from EXFO Service Assurance (formerly Brix Networks). The Brix System consists of distributed test points called Verifiers that communicate with, and are managed by, a centralized server application called BrixWorx.

We installed appliance-based Verifiers, that emulate very-busy, multi-line phones, in the network in locations such as Boston and London. The Java applet on your PC makes a call using the SIP call-signaling protocol to one of these Verifiers. The Verifier answers the call and then measures the quality of the "conversation." BrixWorx assembles all of the test results and provides you with the nifty graphs and tables you see when your test is completed.

Using the unique Tri-Q Analysis, TestYourVoIP.com measures each of the three important quality axes:

• Signaling Quality: Call setup performance
• Delivery Quality: Call stream performance
• Call Quality: Overall voice quality and call experience

Where does the 1-5 quality scale come from?
Voice quality was traditionally reported as a Mean Opinion Score (MOS) on a scale from 1-5 where 1 is the lowest and 5 the highest. Back in the old days, companies would recruit people to listen to test phone calls and rank the quality of those calls — great job, huh? They would then take the average (specifically the mean) opinion of all the listeners and assign that as the call's quality.

Naturally, it's tough to get a bunch of people to sit around and listen to phone calls all day, so here at TestYourVoIP.com we use a software model to calculate the MOS that we report to you. Our software model is based on an international telephony standard called the E-Model — the ITU-T G.107 standard to be exact.

While the theoretical MOS scale tops out at 5.0, practically speaking, you won't get a 5.0 score no matter how good your network connection is. That's because VoIP codecs introduce some amount of quality loss. For example, the maximum MOS score you can achieve with the quality-preserving G.711 codec is 4.4. For the low-bandwidth G.729 codec, the maximum is only 4.2.

Why do I have measurements in both directions?
Your asynchronous broadband connection has more downstream bandwidth than upstream, right? Take the same size file — you could download it more quickly than you could upload it. The same is true all over the Internet. Couple the asymmetric bandwidth with the fact that traffic may travel along a different (asymmetric) path from point A to point B than it does when traveling back from B to A.

It's especially important to capture those differences for VoIP traffic where you have people trying to carry on a two-way conversation.

If you take measurements in both directions, why do you only show one MOS score on the results page?
Sharp eyes!

You've undoubtedly heard the expression that something is "only as strong as the weakest link." It's true for VoIP calls, too. If we're talking and your speech to me takes longer than my speech back to you, we're going to have a bad call. Think about it. I don't realize that you've said anything so I start talking. Just then your speech reaches me and I stop talking. You think I'm interrupting, so you repeat yourself. Talk for more than a few seconds like this and we'll both be annoyed.

So, we report the lowest MOS score. Drill down into the details to identify which side is causing the problem.

What happens when I select the conserve bandwidth and preserve speech quality testing options?
Before a voice signal is sent over the network, it first has to be processed and packetized. This processing phase requires making some tradeoffs between the amount of bandwidth a voice signal requires versus how clear that signal can be. A signal that is highly compressed will require less bandwidth, but may also be less clear. Conversely, one that is compressed less will require more bandwidth but will retain more clarity.

When you select the conserve bandwidth option, we emulate a lower bandwidth G.729 codec for your test call. When you select the preserve speech quality option, we emulate a higher bandwidth G.711 codec for your test call.

What's a codec?
Codec is an acronym for COder/DECoder. A codec is responsible for converting a voice signal into a format suitable for transport and receipt over a network. The codec at the sending end compresses (Codes) the voice signal for transmission over the network. At the receiving end, the codec decompresses (Decodes) the signal for the listener.

There are many voice codecs, each with its own pluses and minuses. Basically, though, the more compression (and therefore less bandwidth) a codec uses, the lower the possible voice quality. Conversely, the less compression (and more bandwidth) a codec users, the higher the voice quality.

What are latency, jitter, and packet loss and how do they affect my calls?
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.

What is ACQ?
ACQ stands for Acceptable Call Quality, a way of measuring the quality of many calls.

ACQ is based on MOS, the 1-5 quality score described above. ACQ is the percentage of some large number of tests that had a MOS score that was acceptable. We use 3.6 as the lowest MOS score we consider acceptable.

ACQ better reflects how people perceive quality than the commonly used average MOS score. Why? Take 1000 test calls. We might see an average MOS of 4.0 for those calls, and an ACQ of 90%. While the average MOS seems fine, 100 test calls, or one out of every ten calls was simply not acceptable. Most users would be distressed to have poor quality on one out of ten calls. It's just human nature.

What is SIP?
Before you can converse with someone, a call must be setup. Both sides need to be able to find and reach one another, consent to talk, and agree how the call is to proceed. Call setup is a pretty complicated process.

Over the years, quite a few signaling (or setup) protocols have been developed. SIP, or the Session Initiation Protocol, is currently popular and used in many VoIP services. That's why we chose SIP for TestYourVoIP.com.

What are the signaling quality results and how do they affect my call quality?
The signaling quality results refer to the time it takes to accomplish the various stages of setting up a call. From the time you pick up the phone to the time the person on the other end picks up their phone and says "Hello" or "What do you want now?" or whatever your friends say when you call -- that's call signaling and setup.

If you pick up the phone and don't hear a dial tone or the call just never connects, or my personal favorite, you dial the number but still hear dial tone -- what do you do? You hang up. Maybe you try again, maybe not. At any rate, it doesn't matter how good the voice clarity would have been or how low the latency was. None of that matters if you can't even make the call.

One thing to remember, even though we're reporting signaling results from TestYourVoIP.com, the ones that really matter are those you get when you try to make a call using your VoIP service provider's infrastructure. After all, you won't be calling us.

So what are these metrics?

• 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 intial call notification request to receiving the first media packet (conversation). Media Delay includes both the call setup delay and post-pickup delay.

What is the VoIP Traceroute Analysis I see in my Detailed Test results?
When you make a call over the Internet or other IP network, the packets in your call don't leave your phone or computer and immediately arrive at your intended destination. Instead, the packets often travel between multiple "hops" - internet routers or other host computers - on the way to their destination. The patent-pending TestYourVoIP Traceroute tells you what path your RTP call packets took through the network and how long it took to arrive at each hop. Specifically, we test and chart the path from the TestYourVoIP Verifier test point you selected back to you.

Sometimes, an intermediate hop does not support traceroute queries. For example, in a typical home network, the router may not respond to the Traceroute queries. In this case, the TestYourVoIP Traceroute analysis will display each hop your call traverses through the Internet and to your broadband provider, but cannot show your home router. Generally, unless your PC has its own public IP address, it will not show up in the Traceroute analysis.

The left hand side of the graph (Y-axis) lists the name of each hop in your call path. The X-axis, measured in milliseconds, displays how long it took to reach each hop. A longer line means it took more time to reach that hop.

What do I use the VoIP Traceroute Analysis for?
The TestYourVoIP Traceroute path and timing information can be very helpful in diagnosing where problems are occurring. If you're having problems with your VoIP or broadband service, be sure to send your TestYourVoIP Traceroute Analysis to your service provider.

So how is TestYourVoIP Traceroute different from the traceroute on my computer?
Traceroute is a utility available on most computers. It tells you what path a data packet takes through the network and how long it takes to arrive at each hop. The big difference? Your typical traceroute function is telling you how data packets are being routed not VoIP packets.

In many of today's networks, VoIP packets are prioritized and routed differently from regular data packets. Using a generic traceroute function, produces incorrect and misleading results.

How come sometimes I have VoIP Traceroute results and sometimes I don't?
These VoIP Traceroute results do not represent my complete call path, what's with that?
A couple of things to keep in mind:

The process of completing a VoIP Traceroute takes some time -- each hop has to be identified, contacted and respond back. It may be the case that you were just too quick and none (or only some) of the final results were in. Give it another 30 seconds and check again.

As we discussed above, some intermediate hops cannot or choose not to respond to our VoIP Traceroute queries. When this happens, we still report as much of the path as we can, but you might see holes or not find a hop that you expect.

How come my web browser is acting weird?
TestYourVoIP.com is developed and tested on the following platforms. If you think your browser is behaving oddly or locking up, make sure you have the latet version of one of these:

Special note to Firefox users: the popular "AdBlock" browser extension hangs Firefox when you try to run the TestYourVoIP tests. We're working on it, but in the meantime, disable AdBlock or try another browser.

I don't have Java installed. What now?
The Java applet we use to TestYourVoIP.com requires Java Virtual Machine (JVM) version 1.4 or higher to be installed. If you don't have a JVM installed, or are running an older version, we'll give you the option to install it before we test your VoIP. The installation takes a couple of minutes, but you only have to do it once.