Is the voice in voice over IP good enough to bet your business on?
That's the make-or-break question for many network executives as they consider the promise of integrated voice/data. And it's one of several key questions we'll try to answer in this four-part series designed to help companies plotting convergence strategies.
Most users and industry watchers say it's relatively easy to achieve toll quality in the lab, but it can be more difficult in a production network. That helps explain why some companies see a future with VoIP everywhere, while others are hesitant to use it on critical links, or for customer-facing applications, such as in call centers or help desks.
"Traditional businesses like ours are . . . on the conservative side. We're not willing to jeopardize our brand name," says Jeff Fountaine, a network analyst with Armstrong World Industries, a Lancaster, Pa., maker of industrial and home flooring and ceiling products. Among his concerns would be a phone order from going awry because of poor sound quality over an IP link.
Is this a do-it-yourself project?
Part 2 of this series
Users hoping SIP's the answer
Part 3 of this series
Answers to your VoIP questions
Part 4 of this series
Still, the potential cost savings are strong enough that Armstrong is willing to give VoIP a shot for certain applications. The company is planning to test an internal campus-to-campus IP telephony deployment before delving deeper into the technology.
In a recent Network World Survey of 250 IT executives, the top perceived drawback for network convergence was the lack of quality-of-service (QoS) assurance on corporate networks. Almost half of those surveyed said that the quality of IP voice was a drawback.
With voice quality being such a sticking point, the trick is to come up with a good way to ascertain whether your network can support toll-quality VoIP. Some experts say there are hard tests and metrics for proving an IP telephony system, while others say the process is more art than science.
Factors that might diminish the quality of an IP phone conversation rarely lie in the actual VoIP gear anymore, says Mike Hommer, manager of lab testing at Miercom, an independent IT testing and consulting firm and a member of the Network World Global Testing Alliance.
"When we started testing VoIP products in 1997, 80% of the metrics we looked at were related to the performance and voice quality. Now that's down to about 10%," he says. "The quality issues - as far as IP voice equipment being able to efficiently encode and decode voice - have become less of a concern."
The issue now is on the network, Hommer says.
"Some people may have no idea how good or bad their network is for supporting real-time protocols like voice," he says.
Network latency is the No. 1 killer of real-time packetized voice traffic, Hommer says. The result of latency is jitter, which can cause an IP voice conversation to break up. Most IP voice products have jitter buffering technology or other technology that smoothes out and reorders voice packets before turning them into audio, but sometimes excessive network latency cannot be overcome.