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I no longer have a landline and rely on a VoIP provider to call/receive calls through the PSTN. It generally works well, but sometimes, end-users complain of low call quality, even when my ADSL line is idle. I ran PingPlotter and did notice some high latency at some hops along the way.

Does someone know of a utility that I could run on my Windows host to monitor the connection to my VoIP provider (through ping/traceroute or some other means) and warn me when the numbers aren't right for good quality calls?

I know about web-based http://myspeed.visualware.com, but I'd rather an application sitting in the icon bar.

Thank you.

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I don't know of a single application within Windows, but there are certain things that you want to be aware of with voip as far as call quality or other connection based issues go and there are separate applications to look for these issues. Some of this you can just use standard ICMP and SNMP monitoring for, other parts you may need to implement some sort of packet capture (via wireshark or logged tcpdump sessions) to look at sip signaling (port 5060). You can also use MTR to test for routing problems.

Latency - high latency will often cause a delay in audio. Jumps in latency, otherwise known as jitter, will cause choppiness and issues where latter parts of the audio stream may arrive before the former parts.

Packet loss - mild packet loss (~5% and under) will just cause some small distortions in the audio. If it consistent, and not just caused from a few random utilization spikes, it will appear as choppy audio. When it gets bad, if the packet loss is in spurts, it will cause large periods of no audio or if it is just a few packets at a time consistently, it will have constant small breaks in the audio stream which will sound sorta like the other side is underwater.

When testing for latency or packet loss there are two important destinations to test to. First test to your ISP's primary DNS server, that will test your last mile for problems. Second, test to your VoIP provider's Session Border Controllers (SBCs).

ALG, SPI, Intrusion Detection - These are firewall settings that will cause random one-way or no-way audio issues as they will step in when they see something they deem as "not safe" or "malshaped" and either block or adjust the packet. This is devastating to voip. You may also see random dropped calls, either because certain signaling is not making it out and after a while when the voice servers do not receive an ACK, they drop the call, or the call will drop at 10, 20, 30 etc minutes due to ALG settings.

When checking for these settings, you will need to check your own router, your ISP's device (unless it is bridged) and your ISP's network.

NAT Traversal - Port triggering or incorrect port forwarding can cause signaling problems that will prevent a phone from registering correctly, pulling a remote config or re-registering at times. This can lead to the phone either losing connectivity after being restarted or just losing connectivity at random times throughout the day.

These will be settings in your own router and possibly your ISP's device if it is not bridged.

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Thanks for the tips on what to watch for. –  OverTheRainbow Aug 24 '11 at 10:27

If you would like to track calls and all the details you can use GnuGK windows version and patch them to one central point.

From your windows you can check all the real time calls using GnuGK/OpenH323gk. It just takes 5 to 30 minutes for serious installation mind.

Hope it helps.

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Thanks for the pointer, but I was looking for a tool to monitor the SIP connection to my VoIP provider, so I know if it's good enough to make/receive calls. –  OverTheRainbow May 23 '11 at 13:29
    
Yes you can use FreeSwitch, and connect your Phone/Softphone to FreeSwitch and setup FreeSwitch to connect 1 VoIP provider or multi providers. And monitor that all in your FreeSwitch. –  YumYumYum May 23 '11 at 13:45
    
Thanks, I'll see how to install and configure FS for that purpose. –  OverTheRainbow Jul 6 '11 at 11:58

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