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Although the terms “Voice over IP” and “IP telephony” are sometimes used interchangeably, they can — and do — mean different things. Technically, Voice over Internet Protocol (or VoIP) is the exchange of voice information in digital form, via TCP/IP network packets rather than the circuit-switched telephone network. IP telephony is generally considered a broader definition of VoIP-based services, and usually involves a communications device that is connected to a data network, such as a PC or telephone set. Like a traditional business telephone system (e.g., Private Branch Exchange or PBX), IP telephony offers typical phone features, such as conferencing, speed dialing, call forwarding, etc. One of the great advantages of VoIP is that the data network (and the money spent building and operating it) can also service the telephone system, possibly eliminating toll charges for certain calls and/or separate voice circuits (or dedicated voice channels on existing data circuits). Because these new phone systems use communications equipment and protocols, phone service is available anywhere a network connection can be obtained. It is now a network application, and is not tied to a particular type of device (such as the phone set at your desk). And because the phone is a network service, integration with software such as e-mail and contact managers is available, raising the value of features such as voice mail and caller-ID. THE NETWORK IS THE PHONE The most important factor in determining the quality of IP telephony service is the performance of the data network it relies on. This includes both the local area network (LAN) and wide area network (WAN). Without speedy, whistle-clean network communications, the quality of an IP telephony system may not be perceived to be as good as the old PBX system. That’s usually an unacceptable outcome. Without delving into many technical details (there are many to consider) the basic requirements can be expressed simply as ensuring that four key measurements of network communications capacity and quality meet (or better yet exceed) general VoIP standards: Bandwidth: Bandwidth is sometimes confused with speed, and although the two are related, perhaps the best analogy is that of a pipe. Bandwidth is the diameter of the pipe, not the length. A wide pipe can be long or short; it will always take a minimum amount of time to go from one end to the other. The wider the pipe (greater bandwidth) the more packets can simultaneously travel across it, but that does not mean that the packets will arrive more quickly. Adequate bandwidth means there is enough network capacity to handle expected call volume without causing congestion — which creates delays as packets wait in line to be transported or simply drop due to overcrowding. Common VoIP requirements are roughly between 25,000 and 80,000 bits per second (or Kbps) of bandwidth for each simultaneous VoIP call. For example, if you need bandwidth for 10 simultaneous calls across a network link, you need about 250 Kbps to 800 Kbps of bandwidth. Network delay: The delay in transmission of a packet from one IP phone to another should not exceed specific values, generally between 50 and 100 milliseconds. The higher the one-way delay the lower the quality of the call. One-way delay can usually be doubled to express “round-trip-delay” (a.k.a. RTD, or latency), and needs to be measured for all possible “end-to-end” links. For example, if your firm has offices in New York, Washington, D.C., Los Angeles and London, it probably won’t be a problem to link between New York and D.C., but the more critical task will be adequate speed to London and Los Angeles. As a general rule of thumb, delay for direct point-to-point dedicated line connections — such as a T-1 — is usually at least 1 millisecond for every 100 miles of distance. So a link traversing 3,000 miles will have a minimum expected one-way delay of 30 milliseconds, or 60 milliseconds of RTD. Depending on the carrier and type of network service used (such as frame relay, multi-protocol label switching (MPLS), virtual private network (VPN)) the latency will differ, sometimes dramatically. Change in network delay: Not only must the network delay be within specification, but the amount that the delay changes must also be minimized. This variation in delay is referred to as jitter, and a high value (indicating wide variation in delay from one moment to the next) can also be detrimental to achieving required call quality. The more the VoIP packets arrive at unpredictable intervals, the greater amount of buffering necessary, because each packet contains a fixed interval of voice data, usually 20 milliseconds. If these jitter buffers get overtaxed, packets get dropped and quality degrades. The specific value required will depend on your implementation. Loss of network packets: Every time a VoIP packet is lost, the system tries to substitute a replacement packet to preserve the rapid-fire order in which voice packets arrive. Of course, the system cannot effectively replace the lost data in a single packet, but given that each VoIP packet usually contains 20 milliseconds of voice sound, you probably won’t be able to detect the difference in a small number of bogus replacement packets. However, when packet loss rises above a certain threshold, the difference becomes noticeable. Packet loss generally should not exceed 1 to 2 percent of the total packets sent or received. Causes for packet losses include insufficient bandwidth, excessive jitter, poor data circuit quality, equipment and/or software problems, etc. Ensuring your data network is ready for phone service provides the foundation for a smooth transition to IP telephony. VOICE OVER IP WITHOUT IP TELEPHONY If you have several PBX systems linked together (say one in each office), depending on the age of your systems they may be able to support a feature known as VoIP trunking. Currently the connection between the PBX systems may be through either dedicated voice channels taken out of your wide area network (WAN) links or circuits dedicated entirely to the PBX systems. Either way this can be expensive, as each voice channel consumes between 56K and 64K of equivalent data bandwidth and enough channels need to be available to handle the expected call volume. For example, if you need to provide for up to five simultaneous voice calls between two offices, five channels will have to be dedicated to the phone systems and taken away from carrying data. If on average only two of the five channels are used, you are paying for unused capacity that can’t be recovered. By using VoIP to link the PBXs, you can eliminate this loss by having the phone switch convert the traffic that used to go on dedicated voice channels to IP traffic on your data WAN. Those five separate voice channels (representing up to 320K of data bandwidth) go away, and with TCP/IP quality of service tuning, you can allow the available bandwidth on the WAN link to float between data and voice without risking call quality. If the link needs to handle more than five calls it can as well because now there is no fixed number of voice channels that are available. In this example of VoIP trunking there is no implementation of IP telephony, but the use of VoIP within a traditional PBX environment can reduce cost and increase capacity. BYPASS THE (PHONE COMPANY) TOLL If you implement intelligent call routing, also known as toll bypass, you may be able to realize savings from long-distance toll charges by having your phone system place a long distance call as a local call. The amount of the savings depends on your call traffic and your office locations, but if you are not already doing this it may be worth looking into. Using VoIP or IP telephony is not necessarily required. For example, if the firm has offices in New York and Los Angeles, the phone system is configured to recognize the area codes associated with calls that are “local” to each area. (212, 718, etc. for New York; 310, 213, etc. for Los Angeles) Calls made from New York to Los Angeles area codes are first routed to the phone system in Los Angeles which in turn dials the number and connects the call back to New York using the WAN link between the two offices. The call from New York is originated in Los Angeles, avoiding a toll charge. RECOMMENDATIONS Adopt a step-by-step approach to minimize risks. Pilot the project in a smaller office or separate workgroup to gain experience and resolve problems. These complex projects require substantial planning. PBX vendors can provide slower adoption by offering a “mixed mode” or hybrid PBX, using conventional phone sets while the firm migrates over time to IP phones. But this approach can generate greater costs, particularly if new phone sets are purchased. As with many technology decisions, there is no right answer for all firms. You must carefully balance your user requirements, risks, and logistical, technical and financial constraints to arrive at the best decision for your firm. Sam Collier is co-founder of Union Square Technology Group, based in New York.

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