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
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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