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What it is

Enterprise applications are changing and becoming more accessible through the web and more integrated with convergence applications like voice, instant messaging and video. Consolidating infrastructure for distributed call centers, remote access for sales people at customer sites, video conferencing coupled with computer based training curriculum and full featured IP telephony solutions with access to enterprise applications at corporate headquarters are just some of the examples of the proliferation of distributed converged communications.

The characteristics of TDM-based telephony that we have come to expect is guaranteed dial tone, "pin drop" voice quality, and a robust suite of features. An impediment that slowed the acceptance of first generation IP telephony was that the network infrastructure had been optimized for store and forward type data applications like email, which had very different quality of service (QoS) requirements as compared to voice traffic. Also, many early IP telephony call manager features were lacking compared to their TDM competitors and were not as reliable.

Latency - is simply measured as the amount of time that it takes a packet to traverse the network from sender to receiver. High latency results in speakers talking over each other as they wait for delayed packets to arrive as a response from a sender. When planning your network infrastructure you need to be aware that there are some delays that you can and can't control. For example, on your private LAN or leased lines between sites, you have the ability to manage and control bandwidth to ensure that voice packets experience minimal latency resulting in high voice quality. However, by definition voice packets over the WAN take unpredictable routes across many
routers and switches all with different store and forward requirements, buffer/queuing mechanisms etc., which results in uncontrollably high latency. Most experts agree that voice packets can sustain 150 to 300 milliseconds of delay before there is a noticeable impact on voice quality.

Extreme switches normally introduce only 8 to 12 milliseconds per switch when forwarding 64byte voice packets, considerably below the voice quality threshold for latency.

Jitter - is a measure of the variation in latency over time. In data-only networks, jitter is normally not measured as long as the packets arrive in a reasonable timeframe at variable rates. In a converged communications network, jitter can have a big impact on the quality of voice applications resulting in truncated sentences or very choppy dialog. It is critical that the network infrastructure has QoS built into the switches to enable them to buffer packets with minimal overhead to ensure smooth packet streaming rates. Jitter rates of less than 1 millisecond is generally considered accepted by most experts. Extreme switches normally introduce a jitter of only 10
microseconds, considerably below the jitter threshold.

Bandwidth availability - bandwidth needs to be available and granularly allocated so that voice conversations are not starved due to congestion on the network. Symptoms of poor bandwidth availability result in dropped packets or out of order packets that need to be resent. The end result is voice clipping, skips and dropped calls.

Extreme switches allow for wire-speed switching at Layer 2 and Layer 3 and all have non-blocking switch fabrics ensuring that the switch backplane will never be a congestion point allowing for maximum bandwidth availability.

Echo - often packet switched voice conversations can be impacted by a reverberation of speech back through the handset and re-transmitted causing distracting echoing. Echo cancellation techniques are being implemented in DSP chips that reside in IP telephones to minimize echo.

Codecs - customer premise equipment (CPE) often comes equipped with voice codec software on the device. The purpose of the codec is to convert analog signals into digital packets often using compression before transmission. There are a number of different codecs ranging from the lightweight, lower voice quality of G.711 to G.729 which is much higher voice quality but has higher bandwidth requirements. Avaya endpoints utilize the highest quality components including codecs and DSPs that support industry standards like SIP and H.323.