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Rich Tehrani

[April 27, 2003]

VoIP: Testing 1,2,3


Editor's Note: This is the eighth in a series of short articles from Rich Tehrani based on recent visits with a number of VoIP vendors. The previous article is available here.

Going out on a limb, I�d say that there are new products coming out in the VoIP space than just about any other technology. Not only are there a slew of new companies in the space -- companies that didn�t exist a few years ago -- but companies from the wireless space, testing space, software market and others are all vying for a piece of the VoIP pie.

If you are designing VoIP systems, you need a way to test them before you deploy. That is where a company like Empirix comes in with its Hammer FX, which can handle IP Centrex, Class 5 features, IP PBXs, unified messaging, IP or TDM conferencing, VoiceXML gateways, voice portals and more.

The latest version of the FX is 1.4 and I had a chance to briefly run through the package and the incredible amount of detailed analysis it can help provide. There are voice quality analysis tools including PESQ and R-Value. Furthermore, there is an enhanced sniffer function that is like a scanning electron microscope for your packets. You can see real-time packets and a ladder diagram. The ability exists to see a decode window that explains what is happening in more detail. Packet metrics and layers are also visible.

Double-clicking on a message brings up all packets associated with a call -- the entire call flow, so you can troubleshoot problems/issues. Obviously this could unleash a tsunami of packets which, by the way, can be shrunk to the more manageable wave-size by way of built-in filters. For example, you can filter based on a SIP To header. The system is also capable of network traversal.

If you are not satisfied with the relatively modest complexity of VoIP packet intricacies, fear not -- you can monitor the SS7 network simultaneously and merge the traces to see a complete view of what is happening. The FX is media aware, allowing you to see all media associated with the call if you like.

By viewing a waveform, you can see interpacket arrival delay via variance off of an expected measure. For example, you could set latency to 20 milliseconds and you would see delays that are greater than this threshold and further, via color coding, you could quickly see when latency increases beyond thresholds. By the way, variance of interpacket arrival times is one definition of jitter.

There is a further filtering mechanism that handles SIP, MGCP, MEGACO, ISDN, SS7, etc. Support is built in for SIP-T (SIP-T provides protocol translation and feature transparency where the PSTN and SIP connect) as well as the ability to search SIP messages and SS7 parameters. The search features are robust, allowing the use of regular expressions when trying to ferret out those elusive problem packets. Of course, as you would expect when you double-click on an error, you can bring back the entire call flow for inspection.

Another sign that VoIP development is on fire at the moment is the extreme interest in TMC�s VoIP Developer Conference. The products and services you deploy are only as good as the testing that�s done before the product is rolled out. You need a good VoIP tester: The Empirix Hammer FX should be on your short list for consideration.

Please talk back to me in our forums.

Rich Tehrani is TMC's president. He welcomes your comments. Participate in our forums.

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