Telephony to Voice Over Wireless LAN.



A complete piece of apparatus for making and receiving calls is called a Telephone Instrument.



Analog and Digital Telephone Instruments.



Analog information-- like sound-- is, in a physical sense, a waveform. Digital information is simply numbers, ones and zeros. Analog phone lines carry waveforms, while digital phone lines carry bits. (A bit is either a zero or a one.)









The conversion of analog to digital then, is the conversion of waveforms to bits. Analog information needs to be represented in binary numbers (1 and 0) in order for the computer to understand. This means that the analog information is divided into sections that are assigned numbers. For example, a continuous wave of a tone would be represented by a large number of squares that, from a distance, resolves into what appears to be a smooth curve, but up close you can see that it isn't really continuous, like a picture in a newspaper. From far away, you see complete pictures, but the closer you look you see that it's just made up of dots. And with the way the mind works, if you get a significant amount of those dots, it looks like a continuous picture.









This holds true for sounds as well. A record is analog data. The voice, or musical sound, goes into an amplifier, which then moves the needle to carve the sound waves in vinyl. When you play back the record, the needle jiggles and is amplified and heard as sound waves.









A CD is digital data. It samples the sound waves into little chunks of amplitude and, in effect, turns the data into ones and zeros. Again, instead of a wave of sound data, you have a line of ones and zeros that mimics the shape of that wave. If there are enough ones and zeros, it's indistinguishable from analog sounds. (Some purists say there is a difference between analog and digital-- analog sounds warmer to them.)Analog phone lines take the audio from your voice, represent it as electromagnetic waves, and then transmit the information through the phone wires to the other end where it's turned back into sound waves. It's still a continuous form of data.Digital phone lines take your analog sound waves, sample them at various rates, and then translate them into bits, ones and zeros, that approximate the wave shape.









Digital information is not a perfect representation of sound data, but if you do a high-enough number of samples, it's indistinguishable from the wave form. One advantage to digital sound information is that you can process the digital information in a number of ways and you can store more data on a smaller amount of space. Most importantly, however, digital information is more resistant to degradation and errors. Analog signals get noisy over time and distance, digital signals are just ones and zeros, so they don't get "fuzzy."



This diagram below shows the difference between standard analog telephone stations and more advanced PBX stations. This diagram shows that analog telephones receive their power directly from the telephone line and digital PBX telephones require a control section that gets its power from the PBX system. Analog telephones also use in-band signaling to sense commands (e.g., ring signals) and to send commands (e.g., send dialed digits). Digital telephones use out-of-band signaling on separate communication lines to transfer their control information (e.g., calling number identification).













KTS-Key Telephone Systems



Often referred to as just KTS, a key telephone system is a premises telephone system that is best known by the phones that have buttons for calling inside an organization and for placing calls outside through the public telephone network. A key telephone system is in the same category as a PBX (private branch exchange), except that key systems rely on the telephone company switching equipment, while PBXs rely on a central control unit located at the customer site. In other words, with a key system, the dial tone is generated at the telephone company central office. A full PBX generates its own dial tones.







Key systems also do not require dialing a number to gain an outside line since all lines are already directly connected to the telephone company central office. On a PBX system, lines are connected to the PBX, and the PBX makes connections to the central office when the outside number is dialed







Designed primarily for small and medium businesses requiring from two to 130 multi-functional telephone sets and/or line combinations. The system resides on the customer's premises and can operate either on its own or in conjunction with a Private Branch Exchange (PBX). A key system or key telephone system is a multiline telephone system typically used in small office environments.







Key systems are noted for their expandability and having individual line selection buttons for each connected phone line, however some features of a PBX such as dialable intercoms may also commonly be present.







Key systems can be built using three principal architectures:



· Electromechanical shared-control



· Electronic shared-control



· Independent keysets







Before the advent of LSIC, key systems were typically built out of the same electromechanical components as larger telephone switching systems. The system was entirely typical and sold for many decades. This system consisted of a central control unit and a number of specialized telephone sets. Each line to the telephone sets was routed using six wires:



· Two wires (one pair) carried the actual telephone line



· Two wires (a second pair) carried control information for that line



· Two wires (a third pair) carried current to a lamp installed at the telephone







A telephone set could contain five, 12, or many individual telephone lines. A common five-line keyphone would be connected using 25 pair cable and an amphenol 50-position micro ribbon connector. The lamps installed at the telephone sets allowed the user to instantly determine the status of all of the individual telephone lines that "appeared" at that set:



· Lamp off — The line is idle



· Lamp steady on — The line is in use for a call



· Lamp flashing slowly — The line is ringing with an incoming call



· Lamp winking fast — A call on the line is "on hold"







A user could select any of the lines simply by pressing the appropriate line button and picking up the handset. A caller could place a call "on hold" by pressing the red "hold" button. This would place the call on hold and then mechanically release the depressed line button, allowing the user to select another line.An individual worker or executive might have a set with one or a few lines "appearing". The system attendent (receptionist) might have a set with many lines appearing so that they could monitor the status of all incoming lines simultaneously.







These systems also supported manual buzzers, intercom lines (with or without selective ringing), music on hold, and other simple features. The features were provided on a line-by-line basis by the selection of particular Key Telephone Units (KTUs) plugged into a pre-wired backplane in the central control unit. The central control unit also provided power for the entire key system (including ringing voltage). A mechanical interrupter in the power supply provided the pulsing voltages for the various lamps, buzzers, and ringers in the system.















Electronic shared-control systems



With the advent of LSI ICs, the same architecture could be implemented much less expensively than was possible using relays. In addition, it was possible to eliminate the many-wire cabling and replace it with much simpler cable similar to (or even identical) with that used by non-key systems.







Additionally, these more-modern systems allowed vastly more features including:



Interactive Voice Response systems, Answering Machine functions ,Remote supervision of the entire system, Automatic call accounting ,Speed dialing, Caller ID Etc.



Features could be added or modified simply using software, allowing easy customization of these systems.







Independent keysets



LSI also allowed smaller systems to distribute the control (and features) into individual telephone sets that don't require any single shared control unit. Generally, these systems are used with a relatively few telephone sets and it is often more difficult to keep the feature set (such as speed-dialing numbers) in synchrony between the various sets.







Small PBX System



This diagram below shows a block diagram of a small PBX system. This diagram shows that a PBX system contains line interface cards that connect the PBX to outside communication lines (such as the PSTN). The PBX also contains station interface cards that adapt the PBX signals to the type of PBX extensions (PBX telephones) that are used with the system. The PBX unit contains a switch to interconnect stations to other stations or to outside lines. PBX call control software coordinates the overall operation of the PBX system. The PBX unit in this example has a data interface connection to allow a control terminal to setup and configure the settings of the PBX system. The PBX also has optional voice mail that connects some of the station interface lines to a voice mail storage system (e.g. a computer hard disk) that allows users to connect to the voice mail and play, transfer and delete stored audio messages. This example also shows that PBX systems may have backup power supplies to allow the PBX telephone system to continue to operate even when the primary power source is lost.







PBX (Private Branch eXchange) is a privately owned telephone switching system for handling multiple telephone lines without having to pay the phone company to lease each line separately.



Normally a telephone line is connected to the phone company's local Central Office through "a trunk." The Central Office is responsible for routing incoming and outgoing calls. It also provides other services like voice mail, call forwarding, caller ID and so on. For this service the phone company receives a monthly fee. A company requiring dozens or even hundreds of phones would quickly incur a very large phone bill!









A PBX essentially takes the place of the phone company's Central Office within the company by acting as the exchange point, routing calls. With a PBX in place, each phone only needs an extension, not a phone number, and the PBX handles all calls made from desk-to-desk within the company.







When an outside call is required, an access number is dialed first. The PBX then transfers the call to the phone company's Central Office. From there the call is routed normally.







A PBX reduces cost because the company only pays for the number of lines liable to be connected at any given time to the outside. If a company has 100 telephones, it's unlikely everyone will be making an outside call at once. Perhaps only 10% will require an outside line at any given time. Therefore the company would lease 10 lines from the phone company rather than 100.PBX systems can be bare bones or feature-rich, depending on what the customer is willing to pay. Voice mail, call forwarding, conferencing, intercoming, and transferring are just some of the options available.In a typical office environment, the PBX system connects multiple incoming phone lines to multiple telephone extensions.







Basic PBX switches do little more than cross-connect these lines. As system price rises, functions are added. Some added features can be provided through software and/or firmware upgrades inside the basic hardware. For other features add-on modules are required





.Usually, the PBX device is a piece of hardware that hangs on a wall or mounts in a rack. Some type of patch panel is included that allows connection to internal and external telephone wires. Sometimes, PBX functionality is provided through software. In this type of system, a personal computer controls system operation and adapter cards and add-on modules provide connectivity. Operation is fairly straightforward. Callers that want to reach someone in the company place their calls from any type of telephone. The call is routed through the Public Switched Telephone Network (PSTN) to company-specific lines leased on a monthly basis from a telephone company.







The PBX system answers the call with a recorded greeting, plays a menu of connection options to the caller, and then routes the call to the appropriate employee extension or to a holding queue (ACD queue or hunt group) for a department, such as sales or support. In installations where the company wants calls answered by a person instead of a machine, the calls are first routed to an operator or receptionist who then forwards the call to the proper extension or department. Calls transferred to an extension will ring at a particular phone, usually a desk phone somewhere in the office. If the extension owner picks up the phone the call is connected. If not, the call is usually transferred to voice mail. When callers know what department they want but don't have the name or extension number for a particular individual, they usually have the option to be sent to a holding queue to wait for the next available agent (employee) to take the call.







Many low-end systems do not offer any type of holding queue, and callers must know who they want to speak with before they call. Other low-end systems send callers to a "hunt group" - a list of phone numbers to try and find someone available. Hunt groups usually have the drawback that every extension number must be tried, in the same order each time, in an attempt to find an employee that can take the call. In such cases, the first extension on the hunt list usually gets swamped with calls while other extensions are used only when there is a heavy load. Another disadvantage of hunt groups is the time it takes to try each extension to find one that isn't busy and has someone ready to pick up the phone.







Higher-end PBX systems employ a variety of techniques to assure that calls to a holding queue are answered more efficiently. The most prevalent approach is through the use of Automatic Call Distribution (ACD) queues. A system with ACD queuing keeps track of which employees are already taking calls and how long it has been since each person finished prior calls. Incoming calls are put into the queue waiting for the next available employee and then routed automatically to the employee that has been off the phone the longest. ACD queuing evenly distributes calls to employees while insuring a minimum wait time for each caller on hold. The ACD queue feature can add considerably to the cost of the PBX system but is often a major factor in customer/caller satisfaction. Serious businesses usually need the advantages of true ACD queuing.





Centrex

Before the advent of VoIP there were, broadly speaking, two ways to provide office workers with telephone service, an organization can install a PBX which switches calls within the office, connects the office phones with the larger telephone network, and delivers services such as voice mail and automated attendant. an organization can contract with the telephone company to provide the same services using the switch installed at the telephone company's premises. This service (telephone company hosted switch) is called Centrex.





Advantages of Centrex over a PBX?

Centrex is offered by a service provider who is responsible for purchasing, installing, maintaining, and operating the necessary equipment...In contrast having a PBX means in essence being your own telephone company. You become responsible for everything although, of course, you can subcontract any or all of these responsibilities. Which is less expensive depends on many factors and requires a careful analysis to determine the approach best for each situation The availability of Centrex provides an exciting new alternative to existing approaches to provide telephone services to organizations professionally.







IP

IP stands for Internet Protocol, the technology that underlies the Internet. And as we all know, the Internet is the most significant revolution in communications technology since the invention of the telephone.







IP telephony

IP telephony, also known as Voice over IP (VoIP), is the use of Internet protocols to carry telephone calls. Previously all telephone calls travelled over wires and circuits dedicated to voice communications. With VoIP, telephone calls are converted into data and then the data travels over circuits along with other data such as email, web traffic, and file transfers.









Understanding VoIP

It is the technique used in transferring telephone (voice) signal over the Internet by using Internet Protocol. It transmits a digitized version of your voice through IP. The digitized voice information is sent out over the Internet just like other data. From the Internet's standpoint there is no difference between other data and digitized voice data unless we somehow identify the data and can recognize the data as a real time signal. Most Internet instant messengers (like Yahoo, Microsoft) allow voice chat. That's one of the types of VoIP. If you have the proper interfacing devices, you can even use an analog phone for chat with these services. While these are technically VoIP, they aren't quite what the world is looking for as a telecom solution. The real world of VoIP is of course, more complex.





Difference between old phone and an IP phone?

The traditional phone we use is an analog phone. It is usually based on the Bellcore standard. Only two wires are needed for analog phones, the wire ends in RJ-11 connectors and setup is as simple as plugging that connector into the wall. An IP Phone is very different, in fact it is a combination of two things, a digital phone and a system that converts the digital information into chunks of information suitable for shipping using the IP. An IP phone requires 8 wires and ends in an RJ-45 connector and unlike the analog phone; setup requires a lot more than just plugging that connector into the wall.







There are many advantages to using IP telephony over traditional approaches to voice communications:



1. The cost of sending data over the Internet is insensitive to distance. An email across the Atlantic costs the same as an email across the office. By converting voice calls into data, VoIP can exploit this distance-insensitive pricing model enjoyed by email.



2. The cost of installing two separate sets of wires in the office, one for voice (the telephone circuits) and one for data (the LAN) becomes redundant. Organisations can reduce costs and improve efficiencies by only needing one communications infrastructure.



3. Until recently the cost of the computing power to switch telephone calls was sufficiently



high that it made sense to put all the intelligence into a central switch (the central office or PBX) and make telephones particularly dumb. Now, however, microprocessors are cheap and phones can be very intelligent. But that intelligence needs an equally flexible infrastructure. Telephone wires cannot provide that.







IP-Telephony Evolution



For more than 15 years, circuit-switched PBX systems co-existed with packet-switched LANs, each one the primary enterprise system platform for a different communication medium – voice and data, respectively, It was not until the late 1990s that circuit-switched PBX and packet-switched Ethernet LAN technologies were merged to fulfill the original promise of digital PBXs. The resulting product solution, commonly referred to as an IP Telephony , utilizes the packet-switched LAN/wide area network (WAN) infrastructure to support traditional PBX-based voice communication and applications.An IP-PBX can be simply defined as a PBX system that supports integrated IP telephony communication capabilities and has the potential to provide customers with several user-productivity and financial benefits. IP telephony is the provisioning of telephony features and applications across packet-switched communications networks using IP standards and specifications. IP telephony lends itself well to individual station user teleworking requirements and customers, with distributed communication requirements, across multiple locations.Many IP telephone models have unique functions and attributes not available with traditional digital telephones, including an integrated Ethernet switch with an associated thin client browser for server-stored information downloads. IP telephones can also be physically relocated across a network without any station user or system manager programming.Significant cost savings are possible using available WAN data network resources for voice communications, if there is spare availability capacity. Additional savings are possible, because more voice calls can be carried over fixed bandwidth carrier facilities if compression-encoding algorithms are used to convert analog voice signals to digital packets in IP format.Just as no two traditional PBX systems are based on the same architecture design, there is no one design standard for IP-PBXs. Many design options can be used to support IP telephony, but a majority of customers prefer a solution that allows them to cost effectively migrate from their existing PBX system while retaining the high performance levels to which they have grown accustomed. Performance-level attributes include both survivability and reliability specifications and availability of system features and functions in support of applications.







Four product migration paths are currently available to PBX customers.



They are:

1.IP-enabled circuit-switched PBX.

2. Pure client/server IP-PBX designs.

3. Converged IP-PBX designs.

4. IP-PBX capping: working behind a circuit-switched PBX







IP-Enabled circuit-switched PBX







IP-enabled PBX allows an enterprise to retain its existing circuit-switched PBX system, with the addition of new hardware interfaces and a generic software upgrade to support IP telephony options, such as IP telephone support, VoIP trunking and distributed port interface cabinets/carriers. This migration option allows an enterprise to extend the life of its installed PBX system, but may prove to be expensive over the long term as more IP station and trunk interfaces are added.Circuit-switched PBX systems originally installed in the late 1980s and throughout the 1990s were not optimally designed for IP peripherals. Media gateway/gatekeeper signalling interface boards required to support IP peripherals are more expensive per interface termination than traditional port circuit cards. For enterprises with small line requirements, this scenario can be expensive when hardware costs are allocated across few installed IP ports.Enterprises with significant IP port requirements will have significant common equipment overhead costs, because the system design requires the same proprietary, expensive cabinet frames and port carriers traditionally used for non-IP ports, in addition to the costly gateway/gatekeeper signalling interface boards.







Pure client/server IP-PBX







A pure client/server IP-PBX requires a business to replace the existing circuit switched PBX common equipment hardware entirely, along with all desktop voice terminals, unless customised IP adapters (gateways) are available for installed analog and/or digital telephones. The pure client/server IPPBX design is optimised for IP peripherals, but can be expensive if a significant number of non-IP ports are needed, such as analog telephones, modems, fax terminals, and PSTN trunk circuits for local central office connections.



Since most of the system voice traffic in this design platform is carried over an Ethernet LAN or IP WAN, it is vitally important that all the infrastructure switches and routers are capable of supporting the most efficient and effective quality of service (QoS) programming options in support of voice communications.







Converged IP-PBX







A converged IP-PBX system seamlessly integrates the call processing, voice switching and port interface attributes of both circuit-switched PBXs and pure client/server IP-PBXs into a unified system design. The ideal architecture design option to support call processing and system operation requirements is a centralised call telephony server used for gatekeeping, call control signalling, dialled digit interpretation and analysis, call routing and software feature provisioning.The ideal converged IP-PBX system design should support direct call control signalling transmission between the call telephony server and LAN-connected IP voice terminals, without intermediary hardware interface equipment. Only rare installations consist of 100% IP ports therefore it is necessary to support interfaces to non-IP station equipment and PSTN trunk carrier facilities. Several design methods can be deployed to support non-IP peripheral interfaces such as desktop interface modules for station equipment, port-limited carrier hubs, full-size port cabinets or carriers similar to circuit switched PBX system common equipment and module interface boards integrated into the call telephony server carrier.







IP PBX Capping



The fourth migration scenario for enterprises that require IP telephony service capabilities is the installation of a small IP-PBX system to work behind an installed circuit-switched PBX. This scenario may be preferable in several situations. One is where the enterprise has an installed PBX system that is unable to support an IP telephony upgrade, or if the upgrade cost is excessive. Another is one where there is a significant financial investment in the existing PBX system (common equipment and voice terminal equipment) and little desire to disrupt system operations, yet also the desire to test the IP telephony waters in anticipation of future mass migration. Networking a small IP-PBX system behind an existing circuit-switched PBX system proved to be a popular solution in the early years of IP telephony. Although a pure client/server IP-PBX may appear to be the most efficient design for a small system trial, a more robust converged IP-PBX system may be the optimal long term solution, because it better facilitates the gradual migration of non-IP ports between existing and the new systems over time.







An IP-PBX And CT Server , Smooth Internet Telephony Transition

Traditional PBX vendors such as NEC, PC-PBX vendors such as Comdial, and some newcomers such as Selsius (now part of Cisco) have already demonstrated — and begun selling — either pure IP-PBXs or IP-enabled PBXs that connect to H.323-ready Ethernet phones. These phones look and act like the standard, proprietary, digital, feature phone that's already on your desk.



What has driven the need for an IP-PBX? Companies like have been selling IPT to the enterprise as an adjunct to existing PBXs, taking advantage of intranets that had been set up between headquarters and remote offices before IPT was even a dream. The installed PBXs work, and work well, so they are not going away. Enterprises can, however, take advantage of rate arbitrage opportunities by installing IPT gateways in each enterprise location to send both voice and fax calls over the intranet to their own locations.







Fax is a big driver for IPT in the enterprise. About 70 million fax machines are in use around the world, many in business environments. These fax machines exist, will continue to be used, and are not IP-enabled. Approximately 40 percent of most corporations' telephone charges come from the cost of sending faxes. There is still a need to use these fax machines as endpoints, and there is also a need to send real-time faxes (T.38 is the real-time fax standard for H.323 environments). So, why the need for an IP-PBX?







It comes back to the infrastructure and network management cost savings. An IP-PBX requires only one wire to the desktop. Since companies already need to install Ethernet for computer connectivity, the MIS department can enjoy tremendous cost savings if it doesn't have to install a separate voice infrastructure. Also, when Ethernet is the only infrastructure needed, system management is much easier. The MIS department only needs to manage the data infrastructure — which can now carry both voice and fax.While the PBX adjunct system certainly fills a large enterprise market need — especially for Fortune-2000-sized companies — there is still a tremendous need it doesn't fill. The IP-PBX is a great way for these companies to grow their infrastructure and add capacity Say you are the MIS director of a company expanding into a new building.







An IP-enabled PBX would certainly be less expensive, from both an equipment perspective (the PBX and the cost of each proprietary digital feature phone) and an infrastructure perspective. Using this reasoning, small to medium-sized companies could start out with only an IP-enabled PBX. While each of the IP-enabled PBX vendors mentioned earlier has H.323 phones to sell with its PBX, H.323 fax machines are not yet available — although they are on the way. When H.323 fax machines enter the marketplace, IP-only PBXs may be able to replace IP-enabled PBXs. the computer-telephone integration (CTI) hardware



(Figure 1). . Enterprise Media Today



Consider the voice and data network needs of a typical company. Most enterprises have a proprietary PBX as the centerpiece of the voice network. There are also usually adjuncts connected to this PBX, typically open systems-based computer telephony (CT) systems. Each of these systems is a standalone server, with its application tied to







Some of these systems may also be connected to the data network. For example, the fax server is connected, since you may be able to send faxes from your desktop computer to a fax machine at another location. Alternatively, the fax server may need to fax back a document (stored on the data network) when a caller request comes into the fax server. If you have an IPT voice/fax gateway, it is also connected to both the voice and data network, bridging the two by converting one transmission medium to the other. Also, most enterprises also have a remote access server (RAS), typically to accommodate employees who need to dial in to the e-mail system, or who need access to a database.







Bringing together all the adjunct servers into one system (which is not as hard as it might seem, since these servers are typically based on open systems), makes it much easier to create applications and manage resources (Figure 2). Applications can focus on their differentiating features instead of having to manage low-level resources. The CT server software takes care of all underlying resource management, and the application has no need to know the various voice inputs and outputs (for example, digital, ISDN, analog, H.323, MGCP). It only needs to know that a call is to be made or received. The type of signaling and the transport mechanism are transparent to the application. The CT server means less work, and much more portability for application providers. For example, a CT server application could automatically move into the IP network world from a pure voice network world if a resource manager layer were there to handle the underlying transport.



Figure 2. Multi-Application CT Server



Now, consider again the IP-enabled PBX scenario. If the CT server and the IP-enabled PBX merge, all the voice needs of the enterprise are run from the CT server, which is now very IP focused. This IP focus does not lessen the importance of the enhanced services expected from the voice side. In fact, these enhanced services are just as important — if not more so — in the IP world, since the compression algorithms make features like play/record, fax, conferencing, DTMF detection, fax tone detection, caller ID (even when coming in from the outside instead of intra-enterprise), and speech recognition even harder. While a pure router gateway has its place, enhanced services are a crucial part of the IP-enabled PBX.







So, if you are an MIS director, think how easy it would be to add a new employee if your company had an IP-enabled CT server. Performing a single directory and administration operation would provide all the phone and remote access services your new employee would need. And if you are a new employee, or working in a new building, turn that phone around to see what kind of connector is on the back.







Hybrid keyphone systems-IP-KTS



Into the 21st century, the distinction between key systems and PBX has become increasingly confusing. Early electronic key systems used dedicated handsets which displayed and allowed access to all connected PSTN lines and stations. The modern key system now supports ISDN, analogue handsets (in addition to its own dedicated handsets - usually digital) as well as a raft of features more traditionally found on larger PBX systems. The fact that they support both analogue and digital signalling types gives rise to the "Hybrid" designation.



The modern key system is usually fully digital (although analogue variants still persist) and with the advent of VOIP, is beginning to embrace this new technology. Indeed, key systems now can be considered to have left their humble roots and are nothing less than a small PBX. Effectively, the only aspects that separate a PBX from a key system is the amount, scope and complexity of the features and facilities offered. Designed to operate in a pure IP networking environment, the IP/KTS benefits from a distributed architecture. Whereas traditional switches are contained within a fixed cabinet, the components of the IP/KTS can be distributed within a network to provide greater installation flexibility and improved network segmentation. Users are able to attach their PC's directly to their IP based telephone keysets and utilise a single Ethernet network connection for both voice and data communications.



Offering connection for up to 70 users per system, the IP/KTS is ideally suited to small to medium sized companies with multiple sites and existing Wide Area Networking (WAN) links. Most WAN links are under-utilised and incur a fixed lease cost per year. By using the IP/KTS and forwarding all voice traffic between offices across these WAN links companies are able to make the most cost effective use of their links and avoid call costs to their telephone network provider.



The IP/KTS is best thought of as a specialized network server offering VOIP (Voice Over Internet Protocol) services to users and allowing data and voice communications to share the same common infrastructure. Sharing the existing data networking infrastructure primarily provides cost savings but also offers several tangible benefits, such as:







· Keyset moves, additions or changes are dynamic and incredibly simple to manage. Often requiring minimal or no configuration changes to the key system.







· System maintenance being possible via a web browser interface from a PC based at a remote location



.



IP Centrex

IP Centrex is a service which provides full-featured telephone service to office workers over the Internet from a switch located at the service provider's facility. The primary difference between traditional Centrex and IP Centrex is that the old-fashioned service required every phone in the office to be connected by separate circuits to the telephone company's central office (CO). This reliance on the wires from the CO limited the number of companies that could offer the service and increased the costs. IP Centrex is delivered over the Internet which eliminates the choke hold the telcos have on traditional Centrex. The result is a more competitively priced offering with more features.Because IP Centrex is not tied to traditional telephone circuits, it is easy to provide service to geographically distributed operations. A company with branch offices can all be part of the same service.IP Centrex does not require the considerable investment in circuits to connect the office phones to the central office and therefore can be offered by many more companies (i.e. increased competition) and at a lower cost (because it does not require the same infrastructure)



IP Centrex is free of the geographical contraints of traditional telephone services making it easy to integrate remote offices, home workers, telecommuters, and indeed travellers in hotels with the office phone system IP phones know their identity (unlike older analog phones) and the maintenance costs associated with office moves is eliminated.Changing offices? Merely take your phone with you and your calls follow! Travelling to New York. Take your phone with you, plug it into the hotel's network and make and receive calls as if you never left the office.







W-PBX



Wireless PBX is Equipment that allows employees or customers within a building or limited area to use wireless handsets connected to an office's Private Branch Exchange system. The various ways this is implemented is explained below.







Wireless access to voice and data services is swiftly becoming the core strategic issue in the telecommunications industry. Equitable and economical access to customers facilitates fast roll out of services and enables profitable operation. Cordless access technologies have been helping to achieve this ideal.







Current issues include:



· the exploitation of microwave point-to-multipoint systems particularly for data and internet applications



· the introduction and/or expansion of Digital Enhanced Cordless Telecommunications (DECT) and other similar technologies as wireless access methods for residential and business subscribers



· the emergence of Wireless Data Centrex, which provides opportunities for many operators to enter the previously sheltered market of data access within buildings, (a key development for European carriers in particular)







DECT Cordless Technology

Introduction

The market is for cordless products is still growing rapidly because users have realized the value of wire-free access. It has already derived many benefits from wireless PBXs and LANs. These include:



· lower cabling costs



· better staff productivity



· reduced phone bills



· greater flexibility



· improved reliability



· integration of wide-area and on-site systems



Customers have become confused over the proliferation of standards and have raised concerns over the security, safety and strength of cordless technology. However, the PBX suppliers do see cordless access as an important part of their 'next generation products'. The real issue is marketing. The major technical issues regarding cordless access technology have been solved. The suppliers who come out on top will be the ones who add value for users by exploiting the intelligence in their products.







DECT was standardized in 1992 by ETSI as a pan-European specification for short-range cordless telephones, wireless PBXs, public access service and wireless local loops. Regulatory authorities regarded DECT as good solution as they were having problems in controlling widely differing types of cordless telephony in the market. Other problems included inadequate security against eavesdropping and the use of diverse and prohibited frequency bands. The DECT Forum claims that DECT is used in more than 110 countries around the world with over 45 million terminals expected to be shipped this year. An installed base of 200 million units is projected by the end of 2005







DECT Standard

The DECT standard (ETS 300 175-1 to 8) primarily defines the parameters for the air interface between a mobile station and a base station. A summary of the standard features follows:







Exhibit 1 - DECT standard features



Transmission Type

Digital



Multiplex Procedure

FDMA/TDMA



Modulation Technique

GFSK



Frequency Range

1880-1900MHz



Carrier Spacing

1.728MHz



Duplex Channels/Carrier

12



Number of Carriers

10



Total Duplex Channels

120



Traffic Density

10,000 Erlang per Square kilometers



Maximum range

300 meters



Mobility Speed

20-50 km/hour









Cordless access will be a crucial element for the next generation of on-site voice and data communications systems. It is of central importance to suppliers in both markets:



· for PBX suppliers who are faced with a saturated market and lengthening product lives



· for LAN suppliers, it has opened the way to a high risk, but potentially very profitable, new market for wireless LANs



A primary incentive for installing wireless technology in offices is reducing the cost of system installation/alteration for organizations that often relocate workers. In the case of voice systems, mobility and productivity of office workers are probably stronger selling points than the long-term cost savings anticipated from wireless LANs.



CT-2 and DECT were once rival cordless technologies in the European market. DECT, however, has replaced most CT-2 installations due to its superior feature set, including support for ISDN services, fast hand-over and enhanced signaling capabilities. The equivalent of DECT in the USA is Personal Access Communications Systems (PACS) whilst Personal Handy Phone System (PHS) is the Japanese equivalent.



DECT technology provides:



· high transmission quality using digital techniques, interrupt-free handovers and ISDN voice



· adequate security against eavesdropping with data encryption



· high data transmission with speeds up to 2Mb/s



· error detection and correction facilities such as CRC, ARQ and FEC (which also leads to enhanced data security)



· comprehensive interworking facilities with other networks: ISDN, X.25, LAN, and GSM



· can be used for voice, data and multimedia in the private, business and public sectors



· operation of picocellular networks in extremely high user densities: 10,000 users per square kilometer possible



· standardized radio interface



· dynamic channel assignment techniques



· self-organizing, thus no frequency planning required



Today, equipment based on DECT is being used in homes, offices, factories and in public places. The first products available were based on Ericsson's proprietary DCT900 technology. The air interface standard, Generic Access Profile (GAP) EN 300 444, is used in association with the common interface standard provided in EN 300 175. ETSI has also developed a standard for DECT access to the Internet.







DECT is known for its flexibility since it can handle virtually any form of telecommunications access. However, competition has arisen in situations where no real alternatives have previously been. For example, it has recently been threatened in the business market by GSM's high capacity 'Office-In-A-Box' solution.















The DECT/GSM Hybrid







Cordless access technologies make very efficient use of airwaves. Such efficiency has resulted in tariffs up to 40% cheaper than cellular mobile. They can handle more than 100,000 users per square kilometer. On the other hand, cellular mobile does provide much wider mobility and as cellular penetration grows this tariff advantage is being lost.



The addition of DECT to a GSM network offers the prospect of extending a network's reach into high-traffic indoor areas such as offices, exhibition centers and shopping malls. Users in these types of environments only require access on the premises and never move faster than at the walking speed. On the other hand, GSM provides continuous wireless access for people who may be traveling at high speeds (such as those in cars or trains), or who may be roaming from one country to another.







Cellular network operators and hardware vendors linked the two technologies in the late 90's. Dual mode handsets were developed which initially search for DECT access before resorting to GSM access. If DECT access is found, a range of virtual PBX services are accessible such as dialing between telephone extensions. Call charges and other costs can be set according to which access is in use at the time. For example, all calls made via DECT access would accrue to the employer's account while all calls made via GSM access could be sent to the customer's personal account. Interest in a dual mode GSM/DECT service culminated in the launch of BT Cellnet's Onephone service in May 1999. However the high cost of the handsets and the ongoing development of 'behind the PABX' GSM offerings led to the closure of the service shortly afterwards.







Table 1: The Future of DECTDECT-ISDN Integration



The integration of ISDN and DECT has resulted in a system that delivers mobility, all the new ISDN features, non-interrupted hand-over ensuring signal clarity, excellent security and high quality of signal transmission. Typical applications include the following:



DECT wireless systems that are adjuncts to new or existing PABX's, with up to 128 cordless phones connecting to the PABX as analogue extensions. These systems support integrated paging and LAN based text messaging to handsets



· A family of ISDN based, wireless PABX systems supporting up to 150 DECT terminals. These can be used as stand-alone wireless PABX's or as adjuncts to larger wired PABX's



· Single line (analogue extension or PSTN) DECT cordless phones



· ISDN compatible corded telephones

















Personal Handy Phone System

Introduction

The Japanese equivalent to DECT is PHS (Personal Handyphone System), however from the outset, it was developed as a public access product, not as a home or business cordless system. This major difference has helped PHS become a successful competitor to conventional cellular mobile phones in Japan. The initial rapid expansion had a lot to do with the unique regulatory and pricing environment of mobile communications in Japan. It was originally developed in response to a need for a digital cordless telephone system. It has been adopted in Japan in two main applications:



· Public Mobile Service;



· Wireless PBX, with possibility of roaming between all three modes.



PHS is also suitable for Wireless Local Loop and high bit-rate data services.



The Japanese PHS market peaked in 1997 at 7.07 million subscribers and then began falling due to pressure from mobile cellular competitors impacting on subscribers. By October 2000 the total number of subscribers in Japan had stabilized at around 5.8 million. Currently, there is some optimism that PHS could have a major come back due to its superior voice quality and higher speed data transmission. The three main Japanese service providers - DDI-P, NTT DoCoMo, and TT Net (previously Astel) - have been aggressively promoting PHS. The superior voice quality of PHS has attracted many corporate customers although younger generations have switched to iMode services in Japan. The data speed of PHS is currently at 128kb/s.







PHS is being deployed in China under the name of Personal Access System (PAS). PAS has already been deployed to around 130-160 cities in China. The number of subscribers is now around 5 million. It has been estimated that the infrastructure deployment cost of PAS in China is US$130 per subscriber. The per minute charge for PAS services in China at just 10% of the cellular mobile charges has been a major stimulant to growth.







In addition, the success of Telecom Asia in promoting PHS over the cellular companies in Thailand has renewed hopes that PHS can succeed in other markets. There are currently around 500,000 PHS users in the greater Bangkok metropolitan area. There are plans that PHS public access systems will also be deployed in Bangladesh and Taiwan.







PHS Technical Standard







The Personal Handyphone System (PHS) standard is a TDD-TDMA based low tier microcellular wireless communications technology operating in the 1880 to 1930 MHz band, now used by millions of subscribers worldwide in public PHS networks, PHS-WLL networks, corporate indoor PBX applications and in the home environment. The cell installation architecture, which uses dynamic channel allocation, requires reduced cell station installation planning and costs for the operators, and can be cost effectively deployed in environments ranging from urban to rural environments.







128 kbs and faster speeds will be available on PHS. The PHS data transmission technology is based on the PIAFS data protocol whose use in existing public networks already occupies 15% of the total network traffic and 45% of the total calls, continuing rapid growth. Users enjoy high speed web browsing, file transfers and convenience of e-mail access anytime, and anywhere. There are currently 15 to 20 manufacturers involved in handset production, with more than 40 models supporting high voice quality Other data transmission applications now popularly adopted include, the location identification service, corporate database access, tele-metering, and handset to handset direct data communication (transceiver).















PHS MOU Group







Not wanting to be left behind by their GSM counterparts, the companies representing interest in the Japanese PHS system established their own MoU Group. Within the group there are 85 odd signatory member organizations including carriers and manufacturers within and outside Japan. These members include MPT of Japan and the telecommunications authorities of Singapore and Australia who joined the group as supervisory agents. Japan's ARIB, Telecommunication Technology Committee (TTC) and Radio Equipment Inspection and Certification Institute (MKK) participated as public organization members and 10 other organizations were initially involved as observers.



The main activities of the PHS MoU include:



· elaboration of PHS technical specifications



· coordination of PHS terminal certification with respect to terminal standards



· promotion of PHS international roaming



· coordination of intellectual properties



· other promotional activities



The marketing push for PHS outside of Japan has resulted in the formation of a number of industry fora in Australia, Indonesia and most recently the United States. The PHS Forum of the Americas now includes fourteen companies who meet regularly to coordinate efforts related to PHS standards definition and promotion of the technology (within South, Central and North America). It also determines approaches to spectrum allocation and provide support for field trials and commercial system testing.







Wireless LANS







A wireless LAN (WLAN) is a flexible data communication network used as an extension to, or an alternative for, a wired LAN in a building. WLANs are useful when employees are on the move, in temporary locations or where cabling may hinder the installation of wired LANs. WLANs may also be used to connect terminals to printers and other devices. The technology avoids the use of costly T1 leased lines often employed in inter-building connections (including WLAN point-of-sale applications such as setting up cash registers in a seasonal display area). They are easy to install, offer the same transmission rates as wired LANs and adequate security. However, the market is widely viewed as vendor-driven and many potential users need to be convinced the products are worthwhile.







According to many industry studies, the majority of WLANs used today are accessed remotely by cellular phones, pagers, etc. The market is still in its infancy and needs to undergo further development, including demonstration of WLAN capabilities to potential users and resolution of spectrum allocation, security and health issues. Complete integration of WLAN with other networks is also very important.



The following exhibits show WLAN specifications for medium and high-speed networks as defined by the Ministry of Posts and Telecommunications in Japan.



Exhibit 2 - Medium Speed WLANs (transmission rates in the range of 256kb/s to 2Mb/s)



Frequency Band:

2.45GHz (ISM band)



Technology:

Spread Spectrum



Antenna Power:

Less than 10mW















Exhibit 3 - Higher Speed WLAN (transmission rates greater than 10Mb/s)



Frequency Band:

18-20GHz (quasi-millimeter wave)



Technology:

Time Division Duplex (TDD) based on 4 FSK or QPSX systems



Antenna power:

Less than 300 mW





WLAN technologies are also being developed in other frequencies such as the 5 GHz band. According to one major computer company, there are many issues affecting the future deployment of WLANs, including 802.11, a standard formally approved in 1996 for WLAN networks. It uses frequency hopping and infrared direct sequencing techniques. WLAN applications are currently prevalent throughout vertical markets, but it is expected that many horizontal applications will follow as 802.11 network infrastructure is installed. 802.11 is expected to make WLANs an economically competitive option for any office environment



Initially WLANs could not deliver more than 1 to 2Mb/s in throughput. In 1999, the IEEE (Institute of Electrical and Electric Engineers) ratified a new standard for high-speed WLANs: IEEE 802.11b. This standard enabled a gross bit rate of 11Mb/s. Apple was the first to adopt this standard, releasing its WLAN system, Airport, in 1999.



The improved performance and lower costs of IEEE 802.11b have led to widespread corporate installations. A large proportion of the installations has been for niche markets to date (i.e.: short-term leased buildings, warehouses, retail outlets, airports, hospitals etc.), however WLANs are now on the verge of breaking into the mass-market.







Competition has already driven down the costs of deploying 802.11b networks from thousands to hundreds of dollars. As a result, 802.11b networks are now being deployed aggressively by businesses to give their employees mobility within the enterprise. Home users are buying cheap 802.11b gear to extend their DSL or cable broadband Internet access wirelessly to the entire house. Major consumer ISPs such as EarthLink are even selling inexpensive 802.11b home gateways.







In addition to homes and enterprises, 802.11b networks are now popping up in public spaces. Users who already have an 802.11b network at home or the office don't need to buy any new equipment to connect. 802.11b public networks use unlicensed frequencies and are cheap to set up and operate, so they will be far less expensive than 3G for the end user. Most importantly, 802.11b public access is available now.Though it is still early, a number of wireless network companies ("microcarriers") are actively building 802.11b networks in public spaces such as hotels, airports, conference centers and retail establishments like Starbucks. They typically strike a deal with a landlord to deploy wireless access points ("APs" - 802.11b wireless transmitter hubs) in the facility and then pay the landlord monthly fees and/or a cut of revenue.An 802.11b AP has a maximum typical range of 500 to 1000 feet. So making 802.11b ubiquitous in all public spaces will require an enormous number of microcarriers.







Short-Range Spread-Spectrum







Deregulation has directly affected short-range, spread-spectrum devices like bar-code readers, point-of-sale systems, wireless PBXs and WLANs. In many countries these systems can now operate with stronger power transmitters without allocation of a frequency license.



Routers or bridges also have to be installed to manage the data flow across a network. WLANS have a much lower cost with only directional antennae interfacing directly with a wireless network access point.



· Exhibit 4 - About spread-spectrum wireless communications



Both frequency hopping and direct sequencing techniques are based on spread-spectrum technology. Spread spectrum technology transmits signals across a range of frequencies using very low energy levels. This means transmissions appear as background 'white noise'. The signal is coded by the transmitter and decoded by the receiver. This makes the spread-spectrum transmissions extremely secure.



Frequency Hopping: During the coded transmission, both transmitter and receiver hop from one frequency to another in synchronization. The rapid movement between frequencies makes it a very secure way of transmitting data. Frequency hopping is best suited to environments where the level of interference is high and the amount of data to be transmitted is low.



Direct Sequencing: This method entails the coded transmission being spread out simultaneously over a wide number of frequencies. The signal is so diffused that it appears as background noise. The receiving station decodes the signal. Direct sequencing is best suited to high-speed, client/server applications where radio interference is minimal.









In-Building Wireless







In-building wireless systems take two forms; those designed for mobility and those designed to replace wires. The market is made up of wireless PBXs (WPBX), WLANs and wireless data collection.







A WPBX enables personal mobility with access to both data and voice. Sales growth for the vast majority of WPBXs is expected to be in stand-alone versions deployed for specialized applications. WPBXs are useful for employees who are periodically away from their desks or stationed in different locations within the same general area such as neighboring conference rooms or site inspections. .







Key findings of a study regarding Wireless Communications Systems (WCS)



As WCS prices decline and performance improves, motivation and purchase justifications will switch. Initially, customers purchase and use WCS to improve employee productivity and communication. However, continued improvement in performance and reductions in price per user, will see WCSs make inroads into markets traditionally monopolized by wired business telephone systems. In short, WCS purchase justifications will increasingly be made on cost. Consequently, this will allow wireless telephones to account for the majority of all business telephones (PBX, Key Telephone and Centrex) in the not too distant future. In fact, already some finely tuned WLAN technologies have become a low-cost alternative serving remote locations. The county of Los Alamos, N.M., opted for wireless LAN technology as both the quickest and cheapest means of linking its seven fire stations, police headquarters and utility departments. It paid off when fires raged and efficient communications were essential. Concerns about noise and interference emerged as a major factor when purchasing and using a WCS.Cellular/Cordless telephone performance and capability would not be acceptable benchmarks for business users. Businesses indicated they require performance and capabilities equal to those of a wired desk telephone. The WCS market will grow according to three price premium plateaus. Initial high-priced premium plateaus will only attract a small segment of the market consisting of those businesses with a strong demand for WCS. This will be followed by a moderate price premium plateau, which will capture the majority of the market. Finally, the remainder of the market will be attracted to WCS only when it reaches parity with the cost of a wired telephone. The need for WCS varies significantly with the type of employee and department within an organization. Needs also vary depending on the size and type of business. WCS needs cannot be satisfied by any one type of wireless phone or system. A Multi-Cell/Multi-User (MC/MU) system will be required for large business facilities (50 or more employees) and a Single-cell/Multi-user (SC/MU) system will be required for small facilities (5 to 50 employees). Various types (differing in size, weight and function) of desktop and portable wireless phones will also be required. The coming two years are very crucial in determining of the Future Integration of Voice, Video and Data all in wireless mode. Once 4G phones are introduced it will be then we will be able to determine the most optimum solution for VoIP and the other features that will come bundled with it.







VoWLAN



It is currently estimated that up to 80% of workers are potentially mobile around their workplace and may have a need to access wireless voice communications onsite. The implementation of wireless LANs (predominantly based on 802.11b and 802.11g standards operating in the 2.4 GHz band) to carry data has lead some vendors to eagerly promote the idea that adding IP enabled wireless handsets to a data network is a relatively simple, inexpensive and reliable method of delivering voice over wireless LAN (VoWLAN). This has lead to declarations that DECT systems, traditionally used for wireless PABX, will not be able to survive in the medium term.







Despite this hype, the consensus of panellists at the Wi-Fi VoIP Futures Summit at the VON trade



show held in Boston in September 2003 was that "…there are a number of challenges that must be met before voice over Wi-Fi goes mainstream….VoWLAN won't see widespread adoption until certain technical hurdles are addressed."1 This paper seeks to dispel some of the hype and highlight some of the limitations potential users should be aware of that are inherent in the 802.11 solutions being offered today.







Wireless Voice – User Requirements



As a starting point, it is useful to summarise the minimum requirements that users in large enterprises demand for wireless voice applications. We will then examine whether these requirements are met by 802.11b/g WLAN standards.· Equipment should be based on uniform industry standards, with interoperability between equipment from all vendors to ensure users are not locked into costly proprietary systems.· A consistently high level of voice quality of service (QoS) is absolutely paramount,particularly with calls to external customers or in critical applications such as healthcare or manufacturing control.



System performance cannot be compromised by interference from other technologies sharing the frequency band or by system load.· Seamless handover of calls on the move between base stations / access points is an essential component of voice QoS.· Radio coverage needs to extend to wherever mobile workers may be. This may include store rooms, canteens, loading docks and outside smoking areas typically not covered by WLAN.



· Users expect their wireless system to be able to be configured to be able to make or receive a mobile phone call on demand.· The wireless network should be totally secure, with encryption of calls considered essential.· Handset performance such as battery life, robustness and the delivery of PABX functionality must match commercial and industrial user expectations.







Ongoing Development of 802.11 Standards



The 802.11 IEEE standard was developed for data, not voice. In fact, standards development is an ongoing process, with no less than 13 different 802.11 standards currently either in use or still to be ratified over the coming years, with a possible 14th standard being recently mooted. Each of these addresses different aspects of WLAN operation and it is fairly common for different manufacturers to produce products that comply with only some of these standards.







802.11 Standards Summary



802.11 b/g do not support features such as security, voice QoS and call handover. This has lead VoWLAN vendors to either leave out these features or to develop their own proprietary solutions which may not be interoperable with other vendor's equipment. "Proprietary systems are plaguing the market on two levels: on vendor specific implementations of VoIP and the VoIP signaling protocols. These protocols would require end users to match their PBX with the Wi-Fi telephony system that they have selected." It's exceedingly difficult to implement multi-vendor wireless voice in the enterprise. Any VoIP phone must work with any provider. On the other hand, DECT is an open standard developed specifically for voice. It has been finalised for several years and the DECT GAP (Generic Access Profile) standard ensures interoperability between different vendors' equipment . DECT has been adopted by over 110 countries worldwide and very importantly, it is the only IMT-2000 family member approved by the ITU for un -coordinated use on an unlicensed spectrum







Quality of Service



Since 802.11 networks were designed to carry data, not voice, 802.11 b/ g standards have no QoS mechanisms built-in to tell the network to p