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More on Telchemy's help with the jitterbugs

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We're continuing our discussion on jitter with a closer look at Telchemy products that predict the quality of voice over IP by focusing on jitter and the resulting packet discards.

As we discussed last time, Telchemy's VQmon software looks at the overall picture for voice quality, but concentrates heavily on the output of the jitter buffers. If the jitter buffer configurations are in good shape, there is a very good chance that a call is going to have a good quality.

But if the jitter buffers are experiencing problems - in particular, if there are many packets being discarded by the jitter buffers - then voice quality will suffer. This is especially true if discards occur in bursts, which are beyond the capability of modern-day codecs to conceal.

One of the significant advantages to Telchemy's approach is that the process is nonintrusive. Monitoring the jitter buffer doesn't have an impact on the information flow. And because it is nonintrusive, the process can continue for every call on an ongoing basis.

This process is particularly effective for finding transient problems. The microbursts of traffic that can wreck a VoIP installation's credibility often disappear by the time an active monitor can be connected to the circuit. The continuous nonintrusive monitoring can see those bursts, measure and record them and alert network administrators to problems that might otherwise go undetected.

The Telchemy products are often embedded in endpoint devices, conceptually residing between the jitter buffer and the codec in order to " see " the output from the jitter buffer. These endpoint devices could be either IP phones or gateways to traditional phones. Inside a network - for testers, probes and analyzers - Telchemy offers an extended product. This network monitoring software adds a jitter buffer emulator which predicts what the operation of the jitter buffers in the endpoints would be on a per-packet basis, as if the agent resided in that endpoint.

Bottom line: Delay variation is one of the most significant parameters, if not the most significant parameter, when it comes to measuring voice quality. It reveals itself in jitter buffer discards. So, analyzing the jitter or its discards and the level of burstiness associated with them and then applying this analysis via a sophisticated, nonintrusive algorithm can accurately predict the actual quality of a VoIP call, in real time, for all calls.

RELATED LINKS

Siemens raises its VoIP
Network World, 09/30/02

Staying connected during a disaster
Network World, 09/30/02

Steve Taylor is President of Distributed Networking Associates and Publisher/Editor-in-Chief of Webtorials.Com. For more detailed information on most of the topics discussed in this newsletter, connect to Webtorials.Com, the first Web site dedicated exclusively to market studies and technology tutorials in the Broadband Packet areas of Frame Relay, ATM, and IP.

Larry Hettick is an independent consultant, with 19 years of experience in telecommunications and data communications marketing and product management for service providers and equipment vendors. He can be reached at larry@larryhettick.com

You can reach the authors at taylor@webtorials.com or larry@larryhettick.com.

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