Sometimes when using VoIP I experience disruptions. I would like to check if the problems could be caused by my internet provider.

How could I best test the quality of my bandwidth (throughput and latency)?

Until now, I have used a script which sends 3600 pings per hour and saves min/max/avg., but I am not sure how representative ICMP packets are.

  • ICMP packets are very unrepresentative as they are often given special treatment by routers which either boost or lower their priority depending on what the marketers want to look good on the sales literature.
    – msw
    Aug 21, 2013 at 21:37
  • I can confirm what msw says. A few years ago, I ran a TCP-based ping test (so sending a SYN and seeing how long the SYN-ACK takes) across an ocean for a few days, and port 5060 (VoIP) was consistently more stable (not faster, just more stable) than for example 443 (web). Which protocol you use likely matters even more than the port.
    – Luc
    Nov 8, 2021 at 12:36

3 Answers 3


MTR is probably the tool you're looking for. I've been using it for a long time and it's helped me troubleshoot a lot of network connectivity problems. It's like traceroute, but it runs continuously and shows you detailed info of every hop along the way.

From the wiki:

MTR relies on ICMP Time Exceeded (type 11, code 0) packets coming back from routers, or ICMP Echo Reply packets when the packets have hit their destination host.

Good luck! VoIP can be a headache to troubleshoot.


The ICSI Netalyzr is a Java application that tests different aspects of your network. It can also check for the buffering behaviour. The drawbacks are that it requires Java enabled in the browser and that it is closed source software.

  • 1
    Another drawback is that it shut down in 2019 and they pulled the Android app from the store as well... Someone seems to have found source code but at the same time writes "the ICIR seem to have no plans to open source or revive Netalzyr". Is this reverse engineered then? I don't know. Not sure if it's functional now or not, but either way you will have to build it yourself and there is no sign of the Android apk. github.com/ifatuus/netalyzr
    – Luc
    Nov 8, 2021 at 12:33

Diagnosing where in the 20+ hops between you and your VoIP partner you're seeing the loss is really difficult because you can't talk to the intermediate devices.

You are testing it in the most direct way, by using your VoIP application, and you've proven that someone is dropping stuff somewhere.

In my experience (debugging carrier networks from the inside) it is most often the case that your ISP has over-subscribed their upstream connection, but they won't tell you if they are seeing loss there, nor can you do much to determine if this is true.

Try changing your VoIP codex to something that uses less bits (hence reducing quality). If your application can use a different IP port number, try that as telcos sometimes throttle VoIP traffic because it competes with their voice services.

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