A good example of this is the local, analog phone line. The frequency range of these lines is limited to between 300 Hz to about 3300 Hz, which is good for one full duplex voice call. With today`s modem technology, we can transmit close to 56k bps on the same pair of wires. However, using the latest voice digitization techniques a single voice call can be nicely converted into less than an 8k bps data signal. Therefore, on the same analog pair of wires and using a 56k bps modem, seven digitized voice calls could theoretically be carried. While there are details and implementation concerns to make this easily work out, the point is that digitized voice and video can be handled much more efficient than analog while the quality to the human ear or eye can be just as good. We next need a good network standard so that everyone can talk to each other across a network. How about using IP-based transmission rules?

Voice must be converted into an IP data stream. This could be done in your PC using a microphone, speaker, and software such as from NetSpeak. However, this may not be the most cost-effective way to accomplish conversion especially for companies with large groups of people or for those without a PC. What is needed is a telephony gateway.

An IP telephony gateway performs two major functions: First, it performs the conversion of analog voice (and video) to and from digital in some code format while packetizing (depacketizing) it. Second, it handles other conversions that may be necessary such as linking different network types - LAN to ISDN line as an example. However, gateways can be very sophisticated and perform many other duties. One very important one is the ability to signal the telephone company`s central office; that is, to request dial tone, hang-up, pass DTMF numbers, and even use SS7 methods. To support a wide variety of analog to digital coding methods - not everyone will be using NetSpeak - gateways must be able to understand and convert the digitized media (voice and video) stream using many formats. In order to support the public, the gateway will have to be able to talk to them, therefore voice recognition and text-to-speech capabilities will be required.

Gateways must be fast. Voice signals are digitized and broken into packets containing 10-50 milliseconds of voice each with 20 to 40 packets per second being generated for a typical call. These must then go through an IP network with 2 to 30 hops (routers) and maybe over strange paths, as in the case of the Internet, with some packets being lost. The International Telecommunications Union standards organization recommends no more than 400 milliseconds of end-to-end delay (latency) for transmitted speech and usually 200ms are fixed propagation delay (speed of electricity through the wires) across the country. This leaves little room for slow gateways and routers

IP Telephony Configurations
Money talks, so voice over IP will become a common way of making phone calls. It already is saving companies big bucks on their International bills. There are several ways this is being done. One is to use the Internet either directly or through an ISP with a gateway. Another is to use specialized Internet Telephony Service Providers (ITSP) who have their own backbone IP network for better quality. A final and somewhat future approach is to use the traditional telephone company. They will have to get into the act or a substantial part of their customer base will go away.

Probably the best approach for a company to take advantage of Voice Over IP (VoIP) telephony is to piggyback IP voice onto their existing data network or virtual private network if it is running IP and has excess capacity. This requires use of a VoIP gateway adjacent to the corporate PBX that is programmed to recognize the phone numbers of branch offices on the corporate network. The PBX sends the identified call to the gateway, where it is digitized, compressed, packetized, and routed over the network. The gateway and PBX at the other end perform the same process in reverse. Some companies program their PBXs to handle "off-net" calls over the IP network. For example, a headquarters initiated call can be routed via the remote gateway to a branch office PBX that redials the customer`s local number and forwards the call. This also saves long distance charges the same way as "on-net" calls do.

Putting gateways everywhere is not cheap; therefore, another approach can be used. This is where the ITSP comes in. They have their own gateways that can be used to enter or exit their networks. The headquarters calls may go through their own gateway in order to compress many calls over a T1 line to the ITSP but individual calls can leave via the provider`s gateway at points local to their destination. A large enough ITSP, like Qwest, will have many local gateways and can offer low rates to corporations that provide large volumes of traffic. However, when ever someone else`s network is being used to carry your company`s traffic one must worry about service levels. Currently, few carriers are giving guarantees of service on VoIP traffic, but this will change.

For the public, there are companies like IDT that have their own gateways but use the Internet as the transmission IP network. They offer good rates but the service may be inconsistent.

Another approach is to use PC software to perform the conversion and packetizing functions. Linked to an ISP, and usually through a directory server, conversations can be carried on with another, similarly configured PC on the other end. Both must also have a microphone and sound card as well as compatible, if not the same, software.

In the near future, traditional telephone companies and expanded ITSPs will offer standard and enhanced IP telephony services. Copright 1999, McGuire Consulting. All rights reserved.

A version of this article with relevant diagrams/illustrations can be found at: http://www.mcguireconsulting.com

jay@mcguireconsulting.com