Use our free VoIP speed test and jitter test analysis tool below to check your Internet quality and to see whether VoIP would be a good option for you. This tool analyses your Internet quality against the key components required for allowing good quality VoIP phone calls. The voice quality of a phone call using VoIP depends on four main connection properties known as Bandwidth, Delay, Packet loss and Jitter (for those who want to know more about these technical terms see descriptions below). These four properties are analyzed and summarized at the end of the test.
If you are experience problems seeing the tool below or when running it, please download the latest version of Java. This tool performs all the tests you need to check out your high speed internet connection, whether you just want to verify your download or upload speeds, or run a jitter test or packet loss test or just checking your connection to verify it is sufficient for quality VoIP service.
Having trouble with Java? We recommend using Java for this test but we do have a NEW HTML5 VoIP test in Beta testing if you wish to try it instead.
This test will take around one minute to complete. The results will include measured information for your Internet connection and consist of a jitter test, upload and download speeds, MOS score and a packet loss test. Use the "Click to start test" button below to start the VoIP test. Ask questions or add a comment about your results here.
|NOTE: The speed test requires JAVA installed on your device to run. Based on your screen size we have determined that you are using a mobile device. Please visit this page from your laptop or PC to utilize this test. Sorry for any inconvience but JAVA is not typically available on mobile devices.|
If your results show five green circles then your Internet connection will support VoIP. If you have four green and then a yellow for consistency of service this is usally fine too. If you are not sure how to proceed, then read the following section on definitions to try to understand what your issue might be.
NOTE: If you see the message "We were unable to measure your connection's jitter/packet loss..." then the most likely issue is that a socket test cannot be initiated (it will do a HTTP/POST test instead of a socket test for speed). This can occur if the ports 20000 and 20001 are blocked for UDP packets (either by something in your setup or by your ISP).
The speed results will often be slower than the speed plan you are purchasing from your Internet provider. The reason for this is because we run an application test rather than a capacity test. In other words we are testing how fast an application can run rathern than physical line speed. Most applications can only run a single connection so it is more accurate to run this kind of test as opposed to a capacity test which runs multiple connections and hides connection delays.
Bandwidth - This is a popular term and you have likely had your Internet provider try to up sell you to a higher "bandwidth" that will give you faster speeds for uploads and downloads to and from the Internet. This is typical displayed in Mbps (Mega bits per second) or Kbps (Kilo bits per second) with home Internet download speeds typically ranging from 3Mbps all the way up to 50Mpbs. it really depends on your needs and how much you are willing to pay every month. Also keep in mind that your upload speed will often be significantly less than your download speed. The good news is that for VoIP you only need around 90kbps worth of bandwidth so if you have a regular high speed connection such as DSL or cable, you should be in good shape.
Delay - If your delay is less than 100 milliseconds, your voice calls should consistently be of high quality. Even delays up to 400 milliseconds (as per ITU) can result in decent call quality.
Packet Loss - Any packet loss up to 5% will likely not be noticed by you when you are making calls. As these are digital packets it is often possible to have a packet loss of 0%.
Jitter - This is measured in milliseconds and is created by some instability in your connection. It is a fluctuation in the signal such that it becomes out of sync or displaced from where it should be in the transmission. It is effectively a continuous variation in the delay of packet delivery. VoIP jitter can be tolerated up to 20ms to 30 ms.
MOS Score - MOS stands for Mean Opinion Score and is actually a score given by a human user when evaluating the quality of voice. As it is an opinion, it is subjective. A MOS score of 4.0 or higher is desired.
SIP ALG - SIP Application Layer gateway is a feature in most routers and is supposed to help SIP based calls when going through your home or business router. Unfortunately it causes more harm than good. Make sure the SIP ALG detector in this test indicates N for NO. If it is Y for YES, try and disable it in your router as per our article on disabling SIP ALG. Then re-test.
Even if you VoIP speed test results are good, you may still run into issues at some point with your voice calls. This can be due to a number of different reasons but is often caused by your internal network not being configured to prioritize the voice packets over all other packets. For example, if you are streaming video while someone else in the household is uploading some pictures to a cloud application, your bandwidth may be consumed by this video and data traffic, leaving very little room for your voice calls to get through. Think of your Internet connection as a pipe and only so much can fit through that pipe at one time. The way around this is to enable Quality of Service (QoS) on a home network router or telephone adapter, and set it to prioritize voice traffic to the Internet. This effectively reserves some room in your pipe such that you will always have room for your phone calls. For more information and help on potential setup, installation, configuration and ongoing issues, please visit our VoIP troubleshooting section.
Don't jump to conclusions and blame your VoIP provider for poor quality of calls as it may actually be an issue with your own home network.
VoIP phone service has become a real option to millions of households in North America with the incredible speeds and reliability provided by modern day Internet service providers. Many home users see savings in excess of $500 per year on their phone bills. This is one great reason why people consider making the switch to VoIP phone service. Did the speed test above indicate that your internet connection was fast enough for VoIP? If so, check out the great deals available using the table on this page.
To find out more about residential (home) VoIP phone service visit our dedicated section to residential VoIP. Here you will find more information about VoIP for home phone users, including educational articles, provider comparison tables, user submitted reviews and more.
The savings don't stop with home phone service. Many businesses in North America are enjoying paying up to 80% less on their monthly phone bills after switching to a VoIP phone service. If your interest is in a VoIP solution for your business then check out our dedicated section to business VoIP. This provides access to many articles and whitepapers that can help with any questions you have, including FAQ's, service features guides and more. You will also have access to provider comparison tables, user submitted reviews and our free price quote service.
#36 : Posted by Blaine Byers on October 25th, 2016:
I ran the HTML5 test 3 times. Each time I got 6 green circles and 1 Red circle. The red was Consistency of Service = 30%, sound is likely to be broken. You do not address COS in your technical terms definitions. I also did not get any SIP/ALG value. Does the COS value mean I would not have good service with VOIP?
-> Response: You need to run the non-html5 test (java test) to get sip alg test results. It is not 100% accurate but if you let me know your modem/router I may be able to help.
Regarding Consistency of Service (CoS), who do you use for your internet and is it wireless by chance? This is a measure of how consistent the speed is over time e.g. if your speed is 3Mbit/s but every 3 seconds it drops to 50kbits/s for 1 second that would be poor for VoIP but the overall average speed would be reasonable. Likely you would experience audio issues during a VoIP call due to lost packets.
#35 : Posted by Paula Lisciotto on August 21st, 2016:
My test shows red for Jitter at 123.78 and red for MOS at 1.29. Yellow for Consistency of Service at 80%. The balance were green. I currently have Hughes Net service but I have not yet upgraded to their Gen 3. Do these scores mean that VOIP service would translate to constant voice break-ups during calls?
-> Response: Unfortunately, yes you could experience audio issues using VoIP. Your MOS score and jitter results are quite poor. Is that HughesNet satellite you are using? I assume so. VoIP over satellite tends to be unreliable I am sorry to say.
#34 : Posted by Fred Stephens on August 14th, 2016:
Hello, on the voip test I received 4 greens, a yellow on consistency of service 66% and a red on MOS 2.4 while the internet is playing a ROKU video. If no roku all is green. Does this mean that I can not use the phone while a ROKU video is playing? I have a cable 10mbs plan.
-> Response: Not necessarily, it just means you need to be aware of this if you start to hear audio issues.
A way around this would be to implement Quality of Service (QoS) on your router to ensure your audio always has highest priority on your bandwidth.
#33 : Posted by Alan Autrand on July 31st, 2016:
I currently have AT&T U-Verse for my land line service. I would like to switch to one of the lower cost VoIP providers to save money. I don't pass your test (3 greens) but have a clear (no static) with only occasional cut out service with U-Verse. Does this mean I might be OK with another VoIP provider or should I upgrade my speed with AT&T?
-> Response: It should be fine, just be aware that any Internet related outages will leave you without a working phone service.
One thing to consider is choosing a provider that has an app for your smart phone (if you have one). That way if you have Internet issues and have good cell service at home, you can still make and receive calls using your home number.
#32 : Posted by Norman Haigh on April 20th, 2016:
I currently have what is apparently called the FWC or Canopy internet service. I am thinking of upgrading to LTE service. My ISP claims that there will be much less packet loss and jitter. Is this likely to be true and will it improve the consistency of service? I would also like to know what settings would be reasonable in configuring quality of service on my wireless routers. I have a Linksys and a Tenda router. I know this isn't much to go on but if you require further info, let me know and I will try to supply it.
-> Response: I don't think the LTE will have a huge impact on the jitter and packet loss numbers but the improved throughput certainly helps. Consistency of service may be better too but latencies through the air and through the cellular providers network are still evident. I frequently use a SIP client running on my iphone and when it is over LTE the quality is excellent.
Regarding router settings check out this article on configuring a router for VoIP.
#31 : Posted by Norman Haigh on March 11th, 2016:
Will increased download and upload speed improve consistency of service, jitter or packet loss?
-> Response: No this would be unlikely, in all honesty.
Consistency of service, jitter and packet loss are parameters that define quality. In fact I would go as far as to say if the speed went up you would likely see more packet loss errors if the quality was already bad.
If you are having issues, try stripping your setup back to a bare minimum and test. In other words, hard wire a laptop direct to the Internet modem and have nothing else connected. Make sure nothing else is running on your laptop but the browser that is running the test. Also if you can repeat this with another device such as a pc I would recommend it in case the laptop has a virus.
Try the same test with your laptop at a friend's house on their Internet service. Hopefully this will narrow down the problem for you. It's a pain but the only way an Internet provider will actually take it serious.
Also try running the tests late at night. It's on a good server but we get hundreds of people testing it each day so you want to rule that out by running it late at night or very early in the morning.
Hope this gives you some ideas.
#30 : Posted by Carol Vollmer on February 23rd, 2016:
Test shows all green except for consistency of service which varies from 25% to 76% from red to yellow. I have set up an Arris SB6141 connected to the OOma Telo and that is connected to the Apple Time Capsule (router storage device) Model A1409. Voice Quality problems mainly at the end I have called. Suggestions to improve appreciated.
-> Response: Can you try connecting your laptop/pc direct to the cable modem and run the test. Disconnect all other devices so nothing else is stealing bandwidth.
Also what speeds are you seeing? I ask because it sounds more like an uplink issue which typically has less bandwidth than downlink for most cable internet service.
Are you seeing any jitter or packet loss when you run the test?
#29 : Posted by David Sackett on January 4th, 2016:
We hear incoming calls very well. When we speak the other person always tells it is very garbled, hearing 1 out of 3 words. Any suggestions?
-> Response: What results do you see when you run our VoIP test?
This sounds like it may be a bandwidth issue on the uplink direction, which often is lower speed than the downlink direction (i.e. outgoing bandwidth lower than incoming bandwidth).
#28 : Posted by John Plosila on November 5th, 2015:
Why does my voice break up when speaking. I hear others Okay.
-> Response: When this happens it normally means you have either low bandwidth on your uplink Internet link or inconsistency of service (or extensive jitter on uplink).
Do you have a decent Internet service? What results did you get when running our VoIP test?
#27 : Posted by Phil on October 21st, 2015:
Can you please tell us where is your server location? Any way I can force a destination to the Voip Test?
-> Response: This server is in Los Angeles, California. Unfortunately there is no way to change this location, as it runs on a dedicated server in Southern California.