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Measuring Voice Quality in VoIP Networks: Going Beyond "Can You Hear Me Now?"

BY LAURA HOLLY
Enterprises everywhere are looking carefully at VoIP these days. And no wonder: the economic advantages of sending voice and data over the same low-cost, IP network can be attractive.
Of course, adopting VoIP means different things to different companies. Some find that bridging their existing PBX systems via IP trunking is a cost-effective way to leverage their IP network and preserve their investment in traditional circuit-switched infrastructure. Others are extending IP telephony throughout the enterprise with software-based call servers and IP phones on the desktop. Additionally, many enterprises are also taking advantage of IP-based audio conferencing and call centers. Smaller companies, eager to offload voice infrastructure maintenance, are turning to provider-hosted solutions, such as IP Centrex or unified messaging services.
But whereas some IT professionals are leaping in with both feet, others are hanging back. And while some are not yet convinced of the VoIP business case, the vast majority is hesitant due to well-founded concerns about voice quality -- especially for calls that will be routed over a WAN or the Internet. Voice is a mission-critical application and many enterprises are simply unwilling to make the leap to VoIP until they can be assured that users will enjoy the same voice quality and call reliability they currently have with their legacy PBX/PSTN infrastructure for each and every call.
IT'S THE NETWORK

While a number of factors can degrade voice clarity, such as poor-quality endpoint equipment or repeated voice signal compression and decompression, by far the number one culprit is the underlying IP network itself. Packet loss, either due to packets dropped by the network or by an endpoint jitter buffer because of excessive network jitter and latency, can significantly degrade the quality of voice transmission. Although voice clarity is an essential indicator of quality, other factors must also be considered. For example, irritating delays caused by excessive network latency can result in parties talking over one another or adopting a walkie-talkie style of conversation. In addition, the call connection experience also impacts users' overall perceptions of call quality. If users don't receive a prompt dial tone when taking a phone off-hook, or if their calls take a long time to connect, fail to connect, or disconnect prematurely, users will be dissatisfied -- even if the voice quality is stellar.
Taken together or separately, any of these problems can make carrying on a normal conversation an exercise in frustration. To make matters worse, these performance factors are not static targets -- the fact that call quality was acceptable yesterday or even ten minutes ago does not ensure good quality on the next call, due to the dynamic nature of IP networks and their ever-changing usage patterns.
MEASURING QUALITY

Given these network realities, how can enterprises ensure both clear voice quality and reliable call connections? While television ads imply that call quality can be assured simply by roaming about with a phone and asking, "Can you hear me now?," the reality is quite different. To meet users' high-quality standards, they must have an effective approach to measuring VoIP network and application performance. This approach must be:
. Pervasive -- Measurement must encompass all IP network infrastructure and services engaged in delivering the VoIP application.
. Continuous -- The VoIP network must be measured continuously, in real-time, and throughout the life cycle of the VoIP application.
. Comprehensive -- Network measurement must include both active performance verification and passive performance monitoring.
Let's review each of these critical network measurement characteristics in more detail.
Pervasive Measurement

The only meaningful measurement approach is one that accurately measures the entire IP infrastructure engaged in delivering the VoIP service. Anything less will result in incomplete data and a misleading analysis of service performance. All of the components that support the voice call must be exercised, including, of course, network, DNS and DHCP service, gateway, gatekeeper, proxy server, or other call and application server performance.
Performing effective measurement requires an enterprise-wide approach, with instrumentation at all service origination or delivery points in the network, such as each branch office or data center. Enterprises deploying VoIP to the desktop should also extend their measurements within locations by measuring call performance directly to endpoint or IP phones at the site.

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