Access the Test Here

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1 VoIP Speed Test Why run the test? Running a VoIP speed test is an effective way to gauge whether your Internet connection is suitable to run a hosted telephone system using VoIP technology. A number of factors must be considered including how many users you will have on a single Internet connection and how many of those users will be on the phone at once. An Internet connection may be suitable to handle one or two concurrent telephone calls, but may not be able to handle five or six concurrent calls, for example. The VoIP speed test considers a wide range of factors that determine your connection's suitability to VoIP. You may also want to refer to our Flexfone Requirements Checklist on how to select a router, calculate the appropriate amount of bandwidth and choose an effective means of implementing Quality of Service or QoS. Especially if selecting our Flexfone Basic option. Running the VoIP Speed Test Click the link below and you will be taken to the Speed Test page. Click on the country and then the Province or State. You will be given optional locations to test, where available. The test requires Java to run and you may be required to download a plug- in for your browser to run or to approve the Java script. Firefox seems to work best. Access the Test Here

2 You will be asked to select Codec from a pull down list Select: G.711 (64Kbps) You will be asked to select number of Lines Unless you have less than 4, Select: 4 Figure 1 What the VoIP Speed Test Looks Like The result will look something like this, with tabs along the side for more details on your performance. It is a good idea to run at different times of the day.

3 Interpreting the Results Now that you have run the VoIP speed test you will need to interpret the results of the test using each of the measures discussed in this article. The following measures are discussed in this article: Bandwidth: Measures how much data your connection can send and receive (expressed as a unit of time). Jitter: Measures how much variation exists between packets (sent and received). Packet Loss: Measures how much information (expressed in packets) is lost during transmission. Quality of Service: Measures how consistent the flow of data (bandwidth) is from your ISP. Mean Opinion Score: This is a numeric measure of the sound quality at the receiving end of the communication circuit. Max Pause: This is a measure of the longest pause recorded during the VoIP speed test (break in audio). Available Bandwidth The first item examined during the VoIP speed test is bandwidth. Bandwidth measures how much data your Internet connection can send (upload) and receive (download) over a time period of one second. In most cases your upload bandwidth and download bandwidth will differ considerably. Upload bandwidth is the limiting factor when determining whether you have enough bandwidth to sustain a certain number of concurrent telephone calls using VoIP. As an example, a typical voice call requires 87 Kbps of upload and download bandwidth. Did your VoIP speed test indicate that you might not have enough bandwidth for your desired number of users? Contact us for suggestions and guidance. Understanding Jitter Jitter is one of the most important factors examined during the VoIP speed test. In basic terms, jitter is the difference between when packets are expected to arrive and when they actually arrive. This often has little impact when you are browsing the web or downloading an e- mail, but for a real- time application like voice over IP, it makes a big difference. In the world of VoIP, timing is everything, and when the timing of packets is being received at unexpected times, an unstable voice connection can result. Watch carefully for this during the VoIP speed test.

4 It's similar to trying to predict when there will be traffic problems on the highway: if you can't predict traffic, you can't be expected to be at work on time in the morning. Some voice providers will implement a jitter buffer which tries to calculate what the maximum amount of jitter will be in a voice conversation, and thus delays the voice conversation in order to keep the conversation from breaking up. A high level of jitter will cause severe degradation in call quality. If you saw a high level of jitter after running the VoIP speed test, you should be aware that your connection might have problems that could prevent it from properly running a VoIP service Packet Loss Packet loss refers to how much information is being lost during transmission; it's expressed in the VoIP speed test results as a percentage. For instance, packet loss of 5% means that 5 per cent of all data transmitted is not reaching its destination. Packet loss can be caused by failures in network cables within the office, excessive network congestion, or general problems with network switching equipment. Packet loss can be constant or occur in bursts. However, bursts are more typical of packet loss. A burst of packet loss over a period of a few seconds will result in little or no voice traffic reaching you and thus you may miss much of the conversation. Packet loss can be caused by a variety of sources. If you noticed packet loss in the results of your VoIP speed test, see below for some suggestions on reducing or preventing it. Avoid network transfers which result in saturation of the upstream bandwidth limit of your internet connection, or enable QoS, or (best option) provide a separate connection dedicated to VoIP Make sure that there are no Ethernet duplex mismatches on your network such as between a cable modem and a router Check network cables between all network devices such as switches, routers, and modems. Replace each one with a new one to make sure this is not causing the problem Restart all network equipment to make sure that your issue is not related to low network resources Quality of Service Quality of Service (QoS) is a major consideration in VoIP implementations and should not be ignored - when running the VoIP speed test, be sure to examine your QoS score! QoS deals with the issue of how to guarantee that packet traffic related to voice will not

5 be delayed or dropped due to interference from other, lower- priority traffic. Imagine that you are uploading a large file to a remote website and at the same time you are trying to talk on the telephone. Your router does not know which type of traffic is more important and thus it treats both types of traffic as the same. This will likely result in a degradation of voice quality. There are numerous ways to handle QoS properly on a network and design your network with voice traffic in mind. QoS is discussed in the following article: VoIP requirements - Quality of Service. A QoS score of lower than 70 per cent means that there is a problem with your connection that will very likely impact voice quality. It is best to run this test mutliple times, especially at times when you are experiencing a voice problem. Mean Opinion Score Rather than judging the quality of a voice connection by subjective terms such as "very bad" or "great," the voice industry has developed a scoring method to quantitatively measure what level of voice conversation you can expect. This is called the Mean Opinion Score or MoS. The MoS gives us an indication of the perceived voice quality of the media after you have received it. MoS is expressed in the VoIP speed test as a number ranging from 1 to 5, where 1 is the worst and 5 is the best. The values do not need to be whole numbers. MoS Ratings Table Rating Definition Description 5 Excellent 4 Good 3 Fair 2 Poor 1 Terrible Excellent sound quality (virtually perfect) Good sound quality resulting in a call similar to a long distance phone call. Phone conversation has some interruptions requiring parties to repeat what was said Each party has issues hearing the other one speak clearly Neither party can communicate effectively

6 Max Pause Max Pause is the longest recorded pause in audio during the length of the VoIP speed test. Ideally you want this to be the smallest number possible, as a large number means that there were pauses in the audio conversation. A long delay could be caused by packet loss or network congestion at the source or destination. Different Results at Different Times Why might you get different results from a VoIP speed test at different times? Networks are dynamic environments and conditions can change depending on network usage, time of day, and other factors. It is best to run the VoIP speed test a few times at different times of the day, or after you experience a network related problem, to try to determine the source of the issue. For instance, you might experience a voice drop- out when an employee is uploading a large file but not at other times. If you didn't have QoS setup properly this would affect your voice conversation and thus it would only appear on a VoIP speed test if the employee was still uploading the file - once they had finished the upload, the test might indicate perfectly good results.