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When law firms adopt voice over Internet protocol technology, they hope that a system allowing them to make telephone calls using a broadband Internet connection instead of a regular telephone line will help them cut costs and integrate applications to serve clients better. But they often overlook some critical details about their data networks that can affect the quality of their calls. Unless a “network readiness” assessment is performed to determine the impact of adding voice calls to the data network, the results of adding voice over IP may be disappointing. Alternatively, upgrading data networks after installing an IP telephone system can result in cost overruns, which can throw the communication budget out of whack. Law firms can make voice calls more reliable over data networks by using methodologies that overcome the possible impairments that can interfere with voice quality. These impairments can disrupt voice conversations unless steps are taken to minimize their impact. Even if your practice relies on an outside vendor or consultant to take care of such things, becoming familiar with these terms will help you interact better with these people. DEALING WITH DELAY Voice conversations are carried over data networks as a series of packets. These packets contain source and destination addresses as well as syllables of speech — all coded as ones and zeros so the information can be understood by the various devices on the network. Delay becomes an issue when these voice packets take too long to arrive at their destination. This situation disrupts the smooth flow of conversation that we have all become accustomed to when using standard telephone service. Data applications like e-mail and file transfers are not affected by delay because the packets are delivered on a best-effort basis, which means that regardless of when data files and e-mails arrive, they can be used or read with no difficulty. But a voice conversation is expected to occur in real time, so delay is a factor that determines the quality of the call. Too much delay will interfere with the natural flow of the conversation, causing frustration and lost productivity at both ends. In client interactions, poor call quality is unacceptable. Delay results from too many packets on the network, causing a general slowdown. This congestion is easily corrected with more bandwidth. A network audit can reveal how much additional bandwidth is needed, particularly on any lower-speed wide-area network (WAN) connections that might be used to link office locations. Firms should also have the routers at each office location examined because they may need to be reconfigured to give voice packets priority handling on the network. For example, if voice and data packets arrive at the router together, but there is not enough free bandwidth to allow them all onto the network, data packets must yield so voice packets can go out first. The routers’ buffers — temporary memory dedicated to organizing traffic before it goes out to the network — may have to be upgraded to store the voice packets for timed release. Timing the release of voice packets reduces the chance of congestion on the network, which ensures a smoother conversation. This mechanism will keep delay within tight limits so that voice conversations are not disrupted. A second problem is that some voice packets may arrive with little delay, followed by those with greater delay. With too much delay variance in the conversation, one party may start talking before he actually knows that it is his turn to speak. The result is that both parties may start talking over each other and have to back off to correct the confusion. This variance in delay is called jitter. For voice communications, it is better to have a consistent amount of delay than varying amounts of delay; the former is predictable and can be dealt with more easily, but the latter is unpredictable and more difficult to overcome. The routers’ buffers compensate for jitter by temporarily storing packets and controlling the speed at which they are offered to the network. But if a router’s buffer is too large, it can cause delay as it waits to fill up with packets. Experienced technicians know how to fine-tune the routers so that they do not contribute to the jitter problem. MINIMIZING PACKET LOSS Sometimes voice and data packets get lost on the network. Other times they are delayed en route to such an extent that the receiving equipment must assume they are lost and request a retransmission. Packets include sequence numbers so that equipment can put them back in the right order before passing them on to the application at the receiving end. If the equipment notices a missing sequence number, it knows to request a retransmission of that packet. Packet loss is not as harmful to voice conversations as delay and jitter, but it can be disruptive if allowed to get out of hand. With e-mail and file transfers, lost packets are simply retransmitted before being put back in the right order. Any voice packets that become lost must be treated differently because no time can be wasted waiting for their retransmission. This would add too much delay to the conversation, so lost voice packets are never retransmitted. The amount of speech lost is usually so small that it goes unnoticed because the human brain is very adept at making sense of the conversation anyway. But if too many voice packets are lost, the result is “clipped” speech, which disrupts the conversation. This can be remedied by adding bandwidth to prevent packet collisions when the network becomes overloaded, or by having the routers rebuild lost packets by examining the surrounding packets that were received, thereby attempting to bridge the gap in syllables through the use of sophisticated speech analysis. NETWORK-READINESS ASSESSMENT Law firms should expect their IP phone systems to be able to do everything their phone systems do today. Getting top performance, however, will require an assessment of the network to determine if it is ready to handle voice traffic. The whole process may take about three days. Two critical items are revealed by the assessment: the number of concurrent voice calls the network can handle without compromising routine data applications, and the quality of the voice conversations. If the quality of the voice conversations falls below accepted industry standards, for example, the network must be reconfigured or upgraded to improve voice handling. The network assessment begins with assembling a profile of the existing network, a process called baselining, which includes examining the kinds of applications running on the network as well as the current traffic volume. A computer with “packet sniffer” software is used to gather this information without disrupting normal business functions or slowing down the network. After a network base line is assembled, a traffic simulator puts the collected base-line data traffic onto the network and adds voice traffic to determine its impact on overall network performance. Peak-hour voice traffic — the time when the most people will be making phone calls — is also accounted for in this assessment. This is determined by looking at the call-detail reports generated by the current phone system. This information is programmed into the traffic simulator to provide a more accurate assessment of a data network’s readiness to handle voice calls. SERVICE QUALITY The traffic simulator might start out by placing 25 voice calls on the local-area network (LAN) that is used to connect computers in your office. As the number of calls increases, voice quality tends to decrease. An optimal balance must be achieved so that both voice and data can coexist on the network without causing disruption to either. Striking this balance involves applying “quality of service” rules to the network during the assessment phase. The objective is to change the behavior of the network so that data and voice traffic are not forced to compete for the available bandwidth. This is accomplished by changing the settings of the LAN switches to classify all the network traffic according to its level of importance. Voice, of course, will have first crack at the bandwidth, as it is a real-time application. Data applications are usually next in line, starting with database access, followed by HTTP for Internet access and FTP for file transfers, with e-mail getting whatever bandwidth is left for best-effort delivery. The performance of e-mail doesn’t really suffer because there is no expectation of real-time delivery among users the way there is with, say, AOL’s Instant Messenger. The routers at each location will also be examined to see how the location handles voice packets over the WAN. This is important because today’s LANs operate at up to 1000 megabits per second (Mbps), whereas the WAN connections between offices are much slower, often working at speeds of less than 1.5 Mbps. So the number of concurrent voice calls possible on the high-speed LAN will be much greater than the number of concurrent calls that can be supported on the lower-speed WAN. As with the LANs at each office location, an assessment is made of the kinds of applications that traverse the WAN, and then simulated voice traffic is added to check call quality against user expectations. Typically, voice traffic on the WAN goes through the network routers of different carriers and Internet service providers, so maintaining consistent voice quality will require the cooperation of these operators. Specifically, their network routers must be set to enforce the same quality-of-service settings that are established on the office routers. If this detail is overlooked, all the changes done in order to run voice on your internal networks will be useless in supporting voice over the WAN. When considering IP telephony, law firms should perform a thorough assessment of their current data network to be sure it can handle the addition of voice traffic. This critical step will head off potential performance problems, provide firms with peace of mind, and enable them to enjoy a successful implementation as they proceed with their investment.
Don Routhier is executive vice president and general manager at TCI, a Springfield, Va.-based provider of converged voice and data networks.

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