Telchemy

Vendor Profile - Telchemy develops technology that enables service providers and major enterprise to deploy and manage Voice over IP, Video over IP, Audio/ Video Streaming and Networked Game services. Telchemy's products provide real time visibility of service quality, estimates of user perceived QoS, and detailed analysis of the root cause of quality degradation for either live calls/ sessions, or for test calls. Their software technology can be integrated by equipment manufacturers into a wide range of products ranging from consumer electronics devices to large telecom systems, monitoring every call and detecting problems in real time; and their OEM products support collection and analysis of real time call quality data, providing easy integration with management systems and other network and call control systems.

Author Profile - Robert Merrill is Telchemy’s Creative Writer. Before joining Telchemy in 2007, he worked for eight years (and surfed a succession of corporate mergers) as an Engineering Technical Writer for BellSouth.net, BellSouth Science and Technology and AT&T Labs. He enjoys writing, learning foreign languages, riding motorcycles, and drinking craft-brewed and imported beer. Robert suggests that the riding of motorcycles and drinking beer, no matter how finely brewed should not be done simultaneously and tasting should only begin after the cycle is put up for the day. Robert writes and works with Dr. Alan Clark, the Founder and CEO of Telchemy and his skills for accurate review of technologies are well reviewed and formidable.

May 21, 2008

P.564 - Putting VoIP Quality Assessment to the Test (By Robert J. Merrill)

Regardless of which topology or performance analysis algorithm is used, a VoIP quality assessment model is useful only if can deliver on its promises by delivering reliably accurate, repeatable results. The ITU-T Recommendation P.564, Conformance testing for voice over IP transmission quality assessment models, is an effort to define operating standards for these models and to present a methodology for objectively measuring and comparing the accuracy of their results.

P.564 establishes minimum criteria for speech quality assessment models that use objective data to assess the impact of IP impairments on one-way listening quality. Originally specific to narrowband (3.1 kHz) telephony applications, the Recommendation was extended to include wideband (7 kHz) telephony in November 2007.

Models that comply with P.564:

• Produce Mean Opinion Scores (MOS) on the ACR Listening Quality Scale---a range of 1 to 5, where 1 represents “Unacceptable” and 5 “Excellent.”

• Evaluate voice quality without regard to the actual voice payload; i.e., independent of the speech content of the analyzed RTP stream.

• Consider the impact of the voice codec used, but do not consider speech level, background noise, delay, sidetone level, echo, or other impairments that can greatly impact the conversational quality of a call.

• Can be deployed in endpoint locations as embedded monitoring agents, at mid-network monitoring locations, or a combination of both.

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February 25, 2008

Accurately Measuring VoIP Performance (By Robert J. Merrill)

As almost anyone who has experienced voice over IP at home or work can attest, VoIP, for all its cost and convenience benefits, is hardly problem-free with respect to reliability and quality of service. If Hum, Crackle, and Pop were the lovable elves of the PSTN, their Orc counterparts—Echo, Garble, and Chop—have proven far less endearing to VoIP users.

In the VoIP world, annoyances take the form of gaps in speech, echo, “robotic” or hollow-sounding speech, clipped speech, and various kinds of distortion and noise. A number of factors can contribute to these problems, but the big three causes are the codec itself and the levels of packet loss and jitter (i.e., excessive variation in the packet arrival interval, which can lead to packets being discarded).

The impairment level introduced by the codec is static, and in most cases negligible. Packet loss and jitter, however, can vary widely and have a number of causes—LAN or access link congestion, route flapping, timing drift, and many other issues related to queuing, bandwidth, etc. Furthermore, the way packet loss and discard are distributed during a call helps determine whether or not the degradation is apparent to the end user.

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