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When you have choppy voice on your line, you should immediately determine the JITTER of your ISP connection to help you in understanding whether it is your LAN or modem that is the problem, your ISP connection, the ISP Network/Backbone, the Primus Network/Backbone, or a general TBB issue. No Jitter may imply that you have a bad phone, bad house wiring (loose cable), or faulty ATA or modem.

The test below will provide your JITTER between your computer and the testyourvoip.com servers - at the particular instance you run the test.

Bad results mean that there is definitely a problem with your ISP connection, and NOT with TBB. This is because this particular test does not interact with the Primus Network or TBB Gateways at all!

Good results imply good quality TBB calls; and should reflect the typical JITTER you will encounter on an actual VoIP call through the ATA to the TBB servers.

In some cases, you may have great results from this test, but you still have terrible TBB Call Quality. This implies that there may be an interconnection problem between your ISP and Primus TBB Gateways, which this test cannot show. You can try and run another test directly to Primus servers as described in Testing your BroadBand connection for TBB. The TBB test does not provide JITTER results, but may help you (and Tech Support) narrow down the issues with your connection.

Simple Steps using your favorite browser to determine the JITTER on your ISP connection:

Note that you should repeat this test a few times when you have problems, and also when you do not. The average of each of the results is a better indicator of your connection JITTER as these values fluctuate every second depending on the traffic on your ISP connection, the Internet in general, and the testyourvoip.com servers! For a comparison over time you can register for free with the testyourvoip.com website and keep ongoing records of your VoIP tests. Registration also allows you to compare your results to those of other users in your area or those using the same ISP (or others). The site caters to American customers and is missing some of the Canadian ISPs.

•Go to URL http://www.testyourvoip.com (Please note that this site appears to currently be offline with no indication if and when there will be a resumption of service.)

Macintosh users should note that the Java compatibility with this site is spotty depending upon which browser you are using to access the website. The Java Embedding Plugin (available through Version Tracker) may solve this problem, but it's a bit complicated to install. It may also introduce minor browser bugs after each use (in my case it messed up typing until I switched to another application and typed something and switched back). Hopefully Apple will update the OS soon to allow the latest versions of Java to be used with browsers other than Safari.

•Select Montreal as the Call Destination and press "Go".

•You can run the test with "Preserve Speech Quality" or "Conserve Bandwidth" or both.

NOTE: Your MOS score cannot be greater than 4.4. See the TestYourVoip FAQ for further info on MOS and other aspects of this test.

•Once the test has completed, select "See Detailed Results" to display the JITTER on your ISP connection. JITTER is listed for each of the Incoming and Outgoing signals with the min, avg, and max millisecond (ms) values.

In general, the max JITTER should be under 50ms. JITTER beyond this will provide progressively worse degradation - choppy VoIP. HIGH JITTER approaching 100ms may make the VoIP service unusable.

There are too many possible variations in results to discuss the issues here, so please post your JITTER results in the MyTBB Forums where someone can help you further.

See Troubleshooting your Internet Connection and ISP on how to sometimes resolve this. Ping and traceroute as described there **may** help show where the problem lies.

If and when you report a JITTER problem to your ISP, they will usually insist that you have a great connection, good signal, and great bandwidth. However, the real issue is congestion which creates high JITTER - the first level rep (CSR) usually does not understand what JITTER is. You may have to call a few times before your link/segment can be upgraded.

In other words, in the case of Cable, they will have to upgrade your neighbourhood network by moving customers to different (new?) cables to spread out the usage. This can take a few months (3-6), and may also depend on how many other customers report the problem!

Similarly, with DSL, they need to add more DSLAM equipment to handle the bottleneck at that point (although this rarely occurs).

Feedback received on this FAQ entry:
  • It might help if you arm yourself with some information when you call into your ISP. You can track your Internet jitter, latency, and packet loss with www.voipspear.com. Do this for a while before you call in and then you can give them information that shows the probvlem.

    2009-08-15 12:49:29 (31366302 See Profile)

  • Hi, I just read your comments regarding high jitters and I've been experiencing this exact same problem for a few weeks already on my VOIP. My ISP is DSL (AT&T). What do you suggest that I can do?

    2009-06-23 20:36:57

  • Hello - I have recently been having issues with my (ISP)service as well as my (VOIP)service "two different co" and have spent quite a bit time trying to resolve QoS issue - Through out this communication with both companies I have realized what a dissadvantage the customer is at unless they can complain with real data and thats where your thread comes in "thanks" - I am posting an addition to your tread I hope you approve http://www.vonage-forum.com/ftopic13039.html this link gets right down to doing there job for them and has a really good Howto link on it with some GNU software "Wireshark" for getting hold of the proof.

    2009-02-04 21:19:47 (thomas007 See Profile)

Expand got feedback?

by canoe See Profile edited by Styvas See Profile
last modified: 2008-02-03 20:23:03