Telephone technology page

    General info

      General technical details of telephone line

      Surprisingly, a telephone is one of the simplest devices you have in your house. It is so simple because the telephone connection to your house has not changed in nearly a century. If you have an antique phone from the 1920s, you could connect it to the wall jack in your house and it would work fine!

      Plain old telephone systems (POTS) telephone line consits of one wire pair which carries full duplex audio and the operating current for the telephone. The telephone connected to line is powered from current limited 48V power source, so phones on-hook, should measure around 48 volts DC. Practically the opearating voltages of telephone systems can vary from 24V to 60V depending on the application, although 48V nominal voltage is the most commonly used. Telephone applications often require and use positive grounding in the central office, where the positive conductor of the 48V power supply is connected to earth ground. The telecommunication industry began the positive ground convention in the 1940s and many telecomm companies still employ the traditionally positive grounded system. The line feeding voltage was selected to be negative to make the electrochemical reactions on the wet telephone wiring to be less harmful. When the wires are at negative potential compared to the ground the metal ions go form the ground to the wire instead of the situation where positive voltage would cause metal from the wire to leave which causes quick corrosion.

      This means that generally when telephone is on-hook, one telephone line wire is quite near to the ground potential and other one carries -48V. When telephone is put off-hook the voltage beween wires going to telephone drops down to the 3 to 9 volt range and typically a current of 20-60 mA will flow through the telephone. The typical operating current range is 20-35 mA. Any more than 55 or 60 mA and it might harm the phones. So the telephone equipment itself does nto need any high voltages to operate. The remaining voltage drop from 48V to 3-9 volts occurs over the copper wire path and in the telephone central electronics. This high voltage is needed in the beginning because the length of the telephone line can be many kilometers, which means lots of wire resistance on the way to drop the voltage.

      The telephone line wires are commonly referred as TIP and RING. Those names originate from plugboard terminology! Originally the 1/4" phone plugs were designed to be used in the telephone plug boards, where the telephone operators manually connected different lines together. The plug tupe as tip-ring-sleeve 1/4" phone plug. The TIP and RING carried the caal signals, and the sleeve could be used for signaling functions or cable shield.

      Typical telephone DC resistance around 180 ohms and AC impedance is typically somewhere around 600 ohms. Typically the telephone central provide from 200 to 400 ohms of series resistance to protect from short circuits and decouple the audio signals.

      Typical telephone line bandwidth is around 180-3200 Hz, which was what phone companies decided years ago be sufficient for speech intelligibility while allowing them to multiplex many calls over coax and twisted pair. The low end is rolled off early to stay away from the mains frequency (50 or 60 Hz) interference and keep telephone transformer size small. The high end cut off is caused by the telephone transmission system (nowadays the audio is digitized at 8 kHz). The high end cut off is critical. Voice on the modern telephone network is digitized at the central office at 8 kHz sampling rate which means that any signal above 4 kHz will be aliased back as noise in the voice band. Most voice CODECs roll off at about -25dB at 4 kHz with a -3dB down point around 3.2 kHz. The phone company uses 8 bit mulaw nonlinear coding which yields about 12 bits of dynamic range. The bottom line is that you can never count on more than about 45 dB signal to noise ratio.

      The typical signal to noise ratio of a telephone line is approximately 45 dB or somewhat less. Telephone line signal to noise ratio is not as easy to quantify because noise comes in many forms, such as electrical interference from fluorescent fixtures or hiss from the many amplifier stages in the voice path. Speech correlated noise can be introduced from non-linear speech coding and compression algorithms. Crosstalk from other conversations is another form of noise.

      Nominal signal level on the telephone lis -9 dBm average speech (at tip/ring). Speech peaks out to +4 dBm are common but will start to clip. The FCC requires that all telephone audio interconnect equipment limit speech to -9dBm, averaged over 3 seconds. Consult FCC Part 68 requirements for all the details.

      Telephone line is balanced voice path. Balanced voice path refers to the fact that the two wires (tip and ring) are only referenced to each other. This design allows signals to travel for miles without expensive shielding by using common mode rejection to remove noise that is induced onto both wires. In a balanced pair of wires, your voice travels on both wires at the same time, but one wire is electrically 180 degrees out of phase with the other. At both ends of the connection (your telephone and the phone company), the signal on one of the wires is inverted and then added with the other (or the voltage difference is otherwise receved). If electrical noise from an appliance leaked on to the wires somewhere between your phone and the phone company. The noise would appear equally and in phase on both wires. When the signal on both wires was inverted and then added together 180 degrees out of phase, the electrical noise would cancel itself out while the voice signal gets stronger. This system allows telephone signals to travel miles on inexpensive twisted pair wires, without significant noise getting into your call. Telephone will work quite acceptably even through untwisted wire pairs.

      Telephone line is full duplex medium. Full duplex means that both people can talk at the same time. In order to send and receive audio through the pair you must use a two wire to four wire hybrid circuit which converts the pair into separate transmit and receive audio paths. The hybrid circuit makes it possible to transmit two channels of information in opposite directions on a single pair of wires. Historically hybrid circuits have used one or two transformers. Bulky and expensive hybrid transformers have been replaced in most telephones by ICs which perform the same function. The hybrid functionality is not perfect. You can get something like 20-30 dB of isolation between signals going to different directions, usually not much more with general purpose "fits for all" hybrids. Usually in normal voice applications this does not cause problems that you hear some of your voice back, but in very long distance calls this can be irrating (in those cases special adaptive echo cancellers are used). Commercial hybrid couplers provide familiar audio connections for full duplex transmit and receive audio. The primary difference between couplers is the amount of trans-hybrid loss or echo from the hybrid. When you send audio into a hybrid, some of the audio leaks back into the receive audio mixed with the caller's voice. The amount of return leakage depends on the type of hybrid and how well it matches the characteristics of the phone line.In some cases there is even intentional leakage. For example the hybrid in the telephone itself id not designe to be perfect. The telephone handset, by design, allows leakage between the send and receive paths ("side tone"). So you can hear back what you talk. Whether it is a transformer or IC, the hybrid must provide at least 1500 volt isolation and surge suppression from lightning strikes of the output of the hybrid is connected to some other equipment. Hybrid systems that are included inside telephones that are completely inside well insulated case (used cannot touch any electronics component in telephone), the hybrid circuti can be a non-isolating type.

      In a telephone system, the biggest contributor to poor audio quality is the handset microphone (it has be be cheap and withstand very hard use) and the limited frequency response of whole system. Within a telephone, the biggest contributor to poor audio quality is the handset microphone. Keep in mind this low cost microphone element is designed to survive years of close proximity spitting and shouting as well as the occasional drop to the floor. The result is a sturdy element that has considerable distortion, a jagged response curve, and substantial dynamic compression. Beyond the microphone, most telephones perform well on a wide variation of telephone line conditions.

      Telephone standards world is fragmented. Typically each country has its own standards because of both the historical roots of the phone service and the desire to protect the local phone market from outside competition. Basically the telephone systems work in the same way in different countries, but there are are some differences which can mean that a devices designed for one country does not meet the regulations of other country and work poorly or not at all. The differences in local technical standards range from minor to severe and affect many of the signaling conditions on local loops. The biggest differences are different wiring practices and connectors, different line impedances, different nominal loop currents, different signaling tones and different electrical safety regulations. Fortunately nowadays many countries are harmonizing many standards across their boundaries so it is nowadays possible to design devices which work well and meet the regulations in more than one country at the time. For example in Europe the European Commission adopted CTR 21 standard covers nonvoice equipments (for example FAX and MODEM) in more than 20 countries. It would be a really good thing if telephone tones were standardised throughout the world. Unfrotunately they are not and will propably never be fully standardized to be same all around world. For example "dial tone", "busy tone" etc. are different in different countries.

      To ring your telephone, the telephone company momentarily applies a 90 VRMS 20 Hz AC signal to the line. Even with a thousand ohms of line resistance, this is still a bit of a shock if you happen to touch the wires, so be careful when you are probing around trying to find a POTS line. Telephone ringivers have some differences between countries.The ring signal is much the same, worldwide. It is around 90V at a frequency between 16 2/3 Hz and 50Hz (20-25 Hz quite common). But its timings are wildly different, as are the return tones it generates.

      Telephone line equipment are designed for specific 600 ohms impedance (or 900 ohms on some places, varies somewhat from country to country) to match the impedance of the line and the equipment on the other line end. A typical standard impedance used in modern telephone line equipment is nowadays 600 ohms (this is used for modems, faxes, etc.). Proper impedance matching is important to get good transmission characteristics especially with long lines. It is well known that, with a signal source of a given impedance, maximum power will be delivered to a load with the same, or matched, impedance. Whether 600 or 900 Ohms is specified, that is a convention. The actual impedance of the system varies somewhat depending on for example what kind of cable is used and there can be frequency dependent variation. For example the actual impedance of 26NL cable, for example, varies from 2000 Ohms at 200 Hz down to 400 Ohms at 4000 Hz. 26H88 loaded cable is around 1200 Ohms at 1000 Hz (loaded cable is not much used nowadays).

      A uniform transmission line has what is called a `characteristic impedance'. This is the impedance that would be measured at the end of such a line if it were infinitely long. The importance of this characteristic impedance lies in the fact that if any length of line is terminated in an impedance of this value, then all the energy flowing along the line is absorbed at the termination and none is reflected back along the line. All energy that is not absorbed by the termination load reflects back on to the copper pair and begins to interfere with the original signal. Because the reflected signal is usually out of phase from the original signal, this starts to cause "common mode rejection" or cancellation of Amplitude at specific "standing wave" frequencies. For the proper operation on long lines, good impedance matching is needed to keep those reflections at minimum. In the real-life telephone subscriber lines the line wire is so short compared to wavelength in the telephone frequencies, that the cables not not have the "true characteristic impedance" on the voice frequencies (few kilometers of able is short line for frequencies below 4 KHz that have 50 km or longer wavelength on cable). The history for 600 ohms is that early telephone system typically used AWG#6 wires spaced 12 inches (305 mm) apart, which made their characteristic impedance exactly 600 ohms at voice frequencies. The 600 ohms imepdance is a general telecom figure, but in is just a simplification of the real situation. In real-life the telephone line is not a purely resistive 600 ohms system. Many countries also describe their own reference model that models the impedance that models the telephone system used in that specific country more accurately than just "600 ohms". The actual models differ depending on some local differences (different central office specifications, different common cable types etc.). The typical voice telephones are designed in such way that thay give good enough match to both the local model and to the 600 ohm reference. Most practical telephone sets do not exhibit a pure 600 Ohm impedance by any stretch of the imagination. If they use a strictly electronic interface, it can be done quite easily. But generally the interface at least on older phones uses a hybrid transformer, and the cost of that transformer is directly related to the impedance vs. frequency characteristics of the telephone set. The return loss of handsets in most countries is specified to be within about 12dB of a defined impedance similar to 600 ohms, which translates to an allowed difference of only a few percent. "Telecommunication System Engineering", 3rd Edition, 1989, by Roger L. Freeman says: "For a conventional two-wire switch, the characteristic impedance is 900 Ohms. This is a compromise impedance, and is the impedance looking into the line circuit. Most equipment to be attached to a two-wire loop is considered to have a 600 Ohm impedance ... Both the 600 Ohm and 900 Ohm values are conventions and are compromises."

      Balanced (for noise rejection) and impedance-matched (for power transfer) transmission lines were clearly necessary for acceptable operationof the early telephone systems, which had no amplifiers. Later, as the telephone network grew, amplifiers, filters and "hybrid" transformers were added to enable long-distance transmission. Proper operation of these components depended critically on rather precise 600 ohms impedances. This 600 ohm impedance is here still nowadays to stay. The real impedance of cables used to way is somewhat different (and varies between different cable types), but still near enough that things planned for 600 ohms will work generally well. Telephone lines should be handled as floating balanced circuits. Any unbalanced device to extract/inject audio to a telephone line has to be coupled in well balanced manner or it will introduce lots of noise to the line. In the telephone central end the two line pair is connected to talk battery and centra office grounds through circuits that have the same impedance to both directions (so it holds the line balancing). On the subscriber end the equipment should connected to the telephone line through an isolation transformer (for example found on modem and properly designed telephone recordign adapters) or the equipment electronics directly connected to line is properly isolated from evetyhting else (many normal telephones have their electronics connected to line directly but they are completely built inside an insulating case and do not have any direct connections outside besides the incoming telephone line).

      One thing to remeber on telephone impedances is that those impedances discussed above are the impedances for AC signals on the telephone voice frequencies (300 Hz to 3.4 kHz). Most telephones are line powered, needing to draw current from the talk battery (on central office) through the line to run their electronics. This involves having a different DC resistance, as opposed to the AC resistance. Normal telephone equipment have typically DC resistance in 100-300 ohms range.

      In a normal telephone volume of the speakers signal that is fed from the telephone set's transmitter to its receiver is significantly less than the volume sent down the line to the distant end. That is, the talkback volume is reduced. Part of the reason is just because we can hear ourselves talk anyway, but the major reason is to equalize the signal loss that the speaker hears to make it similar to the signal loss (over the loop, through the network, to the distant telephone set) that the distant party hears. That loss is assumed to be about 9 dB, or about 1/8th the power level.

      Telephones are designed to work in such way that there can be many telephones on the line, but the system is designed so that only one of them can be in use at the time. If you try to pick up more than one phone at the same time, it might work somehow or not. Tesult of multiple telephone sets being off hook there can be various problems. If there are two phones off hook, they hear each other before that 9 dB of loss over the network. If the two phones are identical, they split that signal as well as the incoming signal (a 3 dB loss). So while the distant party is 3 dB down from normal, the other phone set is 6 dB above normal. This can be very annoying. Because there are more than one phone on the line, those toghether form a different impedance than the system is designed for (normally 600 ohms system becomes termianated to 300 ohms with two 600 ohms phones in use). This can cause impedance matching related problems, which can cause for example problems on your own talkback volume you hear form the phone and sometimes other problems (even feeback noise on some rare cases). The effects of such variation can be strange, because while any one telephone set will have been engineered to be within the limits that would cause problems (such as echo on long distance calls, or even difficulty for people with certain kinds of hearing loss), when two phones are in parallel that may cause some of those effects to be far out of specifications. Then there is more than one phone on the line, they all try to take their operating power from the line. Because the power on the line is limites (lots of resistance on central office and up to kilometers of cable), having many phones at the same time redices the power available to all of the phones (the voltage on the phone drops, current trough every phone lower than normal). Lower than normal power to phone can cause that it does not work properly (for example electronic phones can top working at all work very stangely of do not get neough power). So having more than one phone off-hook at one line essentially a problem of both DC loading and AC impedance. Many phones are simply not designed to be used "in parallel" like this, and methods of coping with this problem and of limiting the current drawn vary widely from country to country. In the UK and many other countries there has earlier been a specification that had to be met if you wanted to allow many phones in parallel. This specification defined the DC characteristic to prevent phones pulling the voltage too low of other phones to operate and also addressed the problem of bell tinkle when pulse dialling. Please note that there are some countries that prohibited the possibility and defined switching schemes that allow only one phone to be connected at a time for "privacy" reasons. Nowadays the chances of successfully connecting phones in parallel depend a lot on when and where your phones were manufactured, it's no longer guaranteed. With some phones you can get graceful degradation when parallel connected, and with some other phones it just does not work that well.

      All subscribers and trunk cable facilities consist of resistance and capacitance. The resistance is determined by the length and gauge of the cable conductors. The cable capacitance is determined by the length of the cable conductors and the spacing between the conductors. The capacitive effect of the cable conductors has a direct relation on the voice band (300 Hz to 3000 Hz) from any given point. The higher the frequency, the greater the loss or attenuation (3000 Hz would be attenuated more than 300 Hz).

      Telephone lines are normally carried through telephone cables that have tenst to hundreds of wire pairs in it. What normally prevents the two signals from interfering with each other is the use of a "balanced circuit" and twisted pairs on the telephone cable. Balanced circuit condiguration combined with twisted pairs (different pairs in wire pair group having different twist rates) gives very good isolation between signals. The ability of that configuration to prevent interference depends on the cable pairs being very well balanced. Any imbalance, and other signals get mixed into your telephone connection. There are three "signals" that are usually strong enough to be detected first, when a cable become unbalanced for whatever reason. The number one is ringing current! The others are 60 Hz from power lines and the clicks from the 48 VDC loop voltage anytime a telephone goes on/off hook or uses pulse dialing. Other indications of an unbalanced line are actually hearing signals from the other lines! For example, the voice caller being able to actually hear the modem tones on the other line. The most likely causes are damaged house wiring, or use of the wrong type of cable for house wiring. Anything that causes the modem lines to be unbalanced will cause them to pick up "crosstalk" from the other lines. Examples would be defective telephone sets, corrosion on terminal, staples through the cable, broken insulation allowing contact with other wires or objects, kinks in the wire, and/or being damp. These are the "six signal killers" (corroded, wet/damp, bent/cinched, insulation problems, impedance/gauge/cable differentials, and bad termination.). Telephone people can measure them individually with suitable test instruments. Corroded wires give effects that fine-tuned transistor/diode testers pick up. Wet and badly terminated wires show up on capacitance. bent, Impedance/gauge and termination show op on inductance. This includes non-twisted lines. Insulation problems show up on current leak tests. All of this can be done with a good line tester; like a specialized multimeter and a few termination blocks and signal generators for the other end. It used to be part of a lineman's set; but now telephone companies usually send specialized crews with suitable special equipment that does all the tests in one go and writes a certification sticker. When problems are seen on measurements, the position where they are located and repair people are sent to repair the cable/connection. The repair work might be anythign from fixing the loose connection to hanging the service to use other wire pair on cable up to cable rapair or pulling a new cable.

      Telephone line work well with just copper wire up to several kilometers without too much attenuation. Nowadays the telephone lines are usually kept below 5 kilometers in length. Standard voice phone calls degrade noticeably when the copper portion of a phone line is longer than 18Kft (6 km) long.

      Loading coils have been eariler (widely tens of years ago) used to extend the range of a local loop for voice applications. Load coils are inserted at specific intervals along the loop (3..6 Kfeet distance). Load coils are inductors that are added in series with the phone line. They compensate for the parallel capacitance of the line, which attenuated the higher voice frequencies more than lower frequencies. By adding inductance (load coils) periodically into the cable facility, the capacitive effect can be cancelled, thus causing the attenuation across the voice band to be equal. Load coils benefit the frequencies in the high end of the voice spectrum at the expense of the frequencies above 4 kHz. Without loading, the frequency response follows an inverse exponetial curve. The longer the line, more high frequencies get attenuated. With loading, the frequency response is essentially flat over the desired range (300 Hz to 3.4 kHz), then drops like a rock. Load coils are are often found at loops extending farther than 12,000 ft. A typical load coil is 88mh coil (type H88), which will cancel 6000 ft of typical telephone cable capacity. They are typically installed at 6000 feet spacing. The typical installation pattern is to have first coil 3Kft from the CO and then after it one coil after every 6Kft. Load coils benefit the normal telephone operation on normal long PSTN line, but do not allow modern broadbans services on lines with load coils. Since ADSL and ISDN depend on frequencies much higher than 4 kHz, they will not work a coil loaded line, because those higher frequencies cannot pass through the coils properly. New digital telephone services require 'unloaded' copper pairs. For example all load coils must be removed for any DSL or ISDN operation.

      While digital telephone lines are quicly coming to the telecom field, it seems that analogue telephone lines are still here to stay for a long time. Strangely enough, fax machines and modems will keep analog lines available even in buildings with ISDN and digital PBXs. Analogue lines still keep going well in the era of digital high speed communications, because you can run both analogue telephone system and high speed ADSL connection over the same telephone wire pair (normal ADSL cannot coexist as nicely more modern systems like ISDN on same wire).

    Business telephone systems

    Business telephone systems very often take use of technologies like PABX, multiline telephones and digital telephones.

      General information on business phone systems

      PBX systems

      The terms "PBX" and "Key" both refer to hardware that enables several telephones to be connected to a smaller number of telephone lines. The term "Key" is was originally used to describe the manual keys or push-buttons on systems like the 1A series Key telephone. The key telephone system is a direct evolution from having multiple phoneson the desk. The earliest system had a series of "keys" (switches) mounted ina box, which allowed one to choose which of several lines was connected tothe phone. One position of the key "hung up" the line while another connectedit to the phone. Usually an intermediate position of the switch allowed one toplace a call on "hold". In early "mechanical" key system such as the 1A1 and 1A2 the user?s phone is actually connected to the line through a "hard" connection (switch contacts) when a line is in use. In older key systems such as the 1A1 and 1A2 six conductors for each line go to each phone on which it appears!Modern key telephone systems have a Key Switching Unit that all lines areconnected to. Standard loop start lines are normally used. All phones in thesystem connect to this KSU. Each phone in a Key system typically has accessto several lines, through the KSU. The phones and KSU work together asfollows. Each phone has an indicator for each line to which it has access.These indicators allow the user to see that state of each line to which they have access. Modern key systems may only require 2 or 4 conductors to each phone.Today?s "Key" system is more like a small PBX with programmable features such as distinctive ringing, hunt groups, and automatic line selection. Key systems allow multiple phones to efficiently share phonecompany lines. Each line has an identity (the phone number) but thetelephones do not. Private Branch Exchange (PBX) is a private telephone network used within a company. Users of such a network share a certain number of external lines for making outside calls. This is typically less expensive than connecting an external telephone line with every telephone in an office. A PBX consists of a switch box and punch block located where the telephone lines come into the building. Usually this PBX system is a device tucked away in an obscure closet together with rest of telephone infrastructure. PBX telephones have an identity of their own. They areextensions with a unique extension number. Each has access to the PBX. ThePBX has trunks to the phone company. The PBX actually switches callsthrough itself based on the users demands, not based on a pre-configuredwiring plan. Station-to-station calls as well as station-to-trunk calls arepossible. Modern business telephone systems can still generally be categorized as Keysystems or PBXs. Note that the advanced features available on many modernKey systems can blur the difference unless one looks carefully.Electronic PBX wiring from PBX to pohones is typically 4 to 8 wires using RJ-11 or RJ-45 modular telephone jacks. These are not usually standard telephone wires (can be similar to normal lines in some systems). Even a line from a simple analog PBX line does not usually look like a standard phone line. The voice path on an analog PBX is typically referred to as a dry pair. Dry refers to the lack of DC current or ring voltage found on regular phone lines.In addition to this there can be another pair that carries the control information and power to telephone. On an electronic PBX telephone, two wires are often used as control lines, which send keypress data to the PBX, and ringer and LED data back to the phone. This control information is required to set up or answer a call. On a digital PBX, your voice is converted to data right in the base of the phone. This kind of specialized digital handsets can run hundreds of dollars, compared to $20 for a standard analog handset.In either case is not possible to use normal POTS telephone or normal telephone recording accessories with PBX lines. You need special phones supplies by PBX manufacturers. If you want to record the calls, you need to do it through the handset cord of your telephone. There are also some PBX systems which provide normal analogue line connection to the lines from the PBX. Network PBXs now generally support standard analog telephones, rather than requiring specialized digital units. Whenever working with PBX system lines, make sure you know what kind of phones are used in this particular system. There are different kind of PBX systems in use. The most wellknown acronyms you might encounter are:

      • PABX (Private Automatic Branch eXchange): is a telephone exchange operated within an organisation, used for switching calls between internal lines and between internal and PSTN lines. A PABX can route calls without manual intervention, based entirely on the number dialed.
      • EPABX (Electronics Private Automatic Branch eXchange): EPABX is a PABX system which is built using electronic controlling and signal switching (to differentiate them from some old relay based designs)
      • PNX (Packet Network Exchange): A communication switching platform that combines PBX and VoIP functionalitues.
      • PMBX (Private Manual Branch eXchange): In some old companies the might still be PMBX, which involves company employed operators manually switching each call using a manual switchboard.
      • KTS (Key Telephone System): KTSs are generally smaller versions of a PBX that provides direct access to CO telephone lines.
      System vendors and users report, however, that the reliability of newer systems is excellent, often reaching the "five nines" level of 99.999 percent uptime. Many vendors recommend running their PBX systems on separate servers, rather than sharing a server with other networked applications. This practice dramatically decreases the potential clash of data and communication systems. Vendors also often provide recommendations for hardware based on their compatibility and reliability testing. Most traditional PBXs were built on proprietary technology, but network PBX solutions take an open-systems approach. At one time, some network systems required specialized cards or handsets, but now network PBX vendors that have fully adopted the IP protocol are actively supporting other industrywide standards. For example, the H.323 standard.Many products also support Primary Rate ISDN (PRI) and standard computer-based telephony interfaces (TAPI and TSAPI). Computer-based PBX systems also tend to support a variety of third-party applications that integrate with the system. Network PBX systems, which are most often sold and installed by systems integrators, consist of a server connected to telephone handsets and incoming telephone trunk lines through a specialized interface device. Calls are managed at the desktop level, typically using a Windows application. These systems, which cost approximately $30,000 to $60,000, are intended for medium and large business applications with hundreds or thousands of users. PC PBX systems are less expensive systems for small-office environments. These systems generally cost a few thousand dollars and add cards and software to an existing server. They typically lack scalability and are best for single-site installations such as branch offices with a couple of dozen to a hundred users.In the PBX environment a line from the CO is called a trunk and a phone is called a line, extension, or station. Many modern electronic PBX systems support Primary Rate ISDN (PRI), which is priced more attractively than other ISDN services. Many EPABX systems have some form of serial interface to communicate with the device itself. In simplest form this can be just an interface to a simple printer that prints out the made calls. In more complicated systems this can allow controlling of the device in more advanced ways. The serial interface used is most often RS232. Usually there is a proprietary protocol (specific to EPABX system) to communicate with EPABX system. So when you interface to EPABX system you need to have software which supports this EPABX systems or you need to develop your custom application for this (to make this you need some specifications of the protocol your EPABX uses). Usually the communication with EPABX is used for extracting Calling Line Identification (CLI) information, Dial Number Information (DNI) and controlling voice speech paths in the EPABX system.There has been a fundamental change in the way an enterprise views aPBX today. Ditto for the position that a PBX commands within an enterprise. Few years ago a PBX was just a voice switch; but today, it can be a switch that can also facilitate data and video communication. However, not all PBXs are like that. With hundreds of choices available in the PBX market, choosing the one that would not onlymeet your current business requirements but also take care of the future needs, is not going to bean easy job. Buy a proven to work platform instead of just a set of boxes. Bet on non-proprietary open standards because they give you more possibilities for future upgrades. Buy a solution that would be inter-operable with applications from a diverse set of vendors and solution providers.Avoid proprietary and closed-architecture based solutions, because additions and changes can be eitherimpossible or very difficult on proprietary and closed standards. At one time, choosing a proprietary standalone PBX phone system was like Russian roulette--make a poor choice, and you could be locked into escalating technical support costs, poor migration paths, and complex system management. Service-level agreements with vendors are also important.PBX field is changing quicly. Many networking people nowadays say that the future of enterprise telephony is clearer than ever: Treaditional circuit-switched PBXs are nearing obsolescence. The makers of IP based telephony equipment say thasto go out and buy a new circuit-switched PBX now is investing in a dead-end technology. IP technology is increasingly becoming the standard for corporate voice communications. But how fast is the question. Should you believe the vendors and embrace it as the newest way for ISPs to increase revenues and for businesses to cut costs, or should you listen to the analysts who say it's still too soon to take the plunge? The answer probably has more to do with your situation than with the technology. These new systems use standard Ethernet connections and deliver more functionality, such as universal messaging. And while many network-based PBX systems may cost nowadays about the same as traditional PBXs on a per-seat basis, prices are plummeting quicly. Network PBX systems may even be more affordable when you also consider the potential system management cost savings, easy integration of remote offices, and reduced long distance charges. With this new technology definately some savings are possible, but the true cost advantages can be difficult to asses.

      Telephone headsets

      Telephone Headset is a device that attachs to the telephone and allows hands-free operation without need to hold the normal telephone handset on one hand. Telephone headsets are commonly used by telephone marketers and callcenter people. Headset allows easy and comfortable talking for long time without taking your hands.The headset industry categorizes headset users into several different categories. The traditional headset user is referred to as "headset dependent" using the headset six to eight hours per day. The next category of user is the "headset intensive" user that spends four to six hours on the phone. The final category is the "occasional" user that may have several protracted calls during the day or blocks of time in the one to three hour time ranges. Headsets typically plug to the telephone line (RK11 connector) or to telephine handset jack (RJ8) through a headset adapter/amplifier module. Those adapters typically have controls for microphone sensitivity and headphone part volume. In interfacing to handset jack please remeber that not all handset connections are not technically similar. Although many telephones appear identical, there are different types of microphone elements that each telephone set utilizes. There are three different categories of microphones being used: carbon, electret, and dynamic. If you have a headset that was made specifically for any of these three technologies, it is not possible to have them perform in conjunction with microphones of another type. Headset manufacturers have to build their products to interface with the phone manufacturer's specifications. Now each manufacturer has application engineers that work exclusively with the telephone manufacturers in order to insure all new phones and current products are properly identified and matched up to work with headsets. This information is provided to all headset suppliers who should be able to initially recommend headsets that work with the phones you are utilizing. Call your supplier to investigate purchasing compatible headsets for your specific phones. Some manufacturers make also general purpose adapters that can be adapted to work with more than one headset type (those have typically microphone amd wiring type selection switches in them). Some modern telephones (PSTN and wireless) have 2.5 mm stereo jack to plug a compatible headset dorectly to them. Typical pinout for 2.5mm stereo plug on the headset unit is the following:

      • TIP:Headphone speaker signal (typically around 32 ohm element)
      • RING: Microphone signal (typically electret capsule)
      • SHIELD: Ground for both speaker and microphone signals
      Note that while this seems to be the most common pinout, this is not the only one used (some manufacturers use different connectors and pinouts).

    Telephone history

    Alexander Graham Bell (1847-1922) is most famous for his invention of the telephone. As a teenager of 18, Bell had been experimenting with the idea of transmitting speech. In 1874, while working on a multiple telegraph he developed the basic ideas for the telephone. He and his assistant Thomas Watson finally proved successful on March 10, 1876, when the first telephone message was transmitted: "Watson, come here; I want you.". This led eventually to the establishment of the Bell Telephone Company, still in existence today, which introduced the telephone to the world.


    The facsimile or fax machine was first invented back in 1842 by a Scottish electrical engineer named Alexander Bain. This was about five years after Morse invented the telegraph.A fax machine electrically breaks up a document into very small pieces, which are called picture elements or pixels and sends them one by one to another fax by way of a phone line. The density of each element is converted to an electrical current which is sent to the receiver. The receiving fax puts the picture elements together as it receives them, until a copy of the original is made.Facsimile (fax) technology, the transmission of images over a telephone line, made its appearance in a commercial application about 70 years ago in the form known as wirephoto, which was used to transmit photographs for publication in newspapers? In the early 1960s, the development of modem technology made facsimile machines practical, although the slow transmission time impeded widespread use. However, in the early 1970s, image data compression was introduced, which drastically reduced transmission time and enabled the fax machine to become an integral part of the business environment. Types Of Facsimiles:

    • GROUP I ( G1 ) / Old FM Transmission time : Approx. 6 minutes per page
    • GROUP II ( G2 ) Transmission time : Approx. 3 minutes per page
    • GROUP III ( G3 ) Transmission time : Less then 1 minute per page
    • GROUP IV ( G4 ) Transmission time : Approx. 10 seconds per page
    The operation of a fax machine is strictly specified by the International Telegraph and Telephone Consultative Committee called "CCITT". This committee sets the standards for all fax equipment thereby allowing different manufactures and faxes in different countries to communicate with each other. Because fax machines are so easy to use (put the paper in the hopper, dial the number, and push the send button), many people assume that the whole process is very simple. The truth of the matter is that communications compatibility between fax devices is complicated and not problem free. Sending a fax is far more complex than just dialing a phone number and sending an image. The calling unit must first confirm that the call is being answered by an actual fax device rather than a data modem, answering machine, or human being. Once it is determined that two fax devices are talking to each other, these devices must then exchange information to find out what capabilities they both support (e.g., data transmission rate, image resolution, data compression schemes, paper size). They then must agree on a mutually supported subset of these capabilities. Next, the phone connection must be evaluated to determine the maximum practical data rate available. Finally, the fax image is sent. However, it is possible that noise or distortion will corrupt the image during its transit of the connecting network. To check for this possibility, the receiving fax device must evaluate the image and send an acknowledge message indicating whether the image was received correctly or whether it had an unacceptable number of errors. The protocol for sending or receiving a fax image and exchanging associated messages is defined in the International Telecommunications Union (ITU) Recommendation T.30. Fax technology continues to evolve, with new features being added on a regular basis, and new fax devices must be compatible not only with the latest models but also with the existing installed base of fax machines. Communications technologies and network structures are changing at a rapid pace (the Internet and cellular phone networks in particular), and all of these must be able to handle fax traffic as well as voice.

    Cordless telephones

    Cordless telephones are one of those minor miracles of modern life -- with a cordless phone, you can talk on the phone while moving freely about your house or in your yard. Cordless phones have many of the same features as standard telephones. The main difference is that cordless phones do not have a cord from the handset to the phone base unit. In cordless telephone this wire is replace with a wireless radio link. A cordless telephone is basically a combination telephone and radio transmitter/receiver.A cordless phone has two major parts: base and handset. The base is attached to the phone jack through a standard phone wire connection, and as far as the phone system is concerned it looks just like a normal phone. The base receives the incoming call (as an electrical signal) through the phone line, converts it to an FM radio signal and then broadcasts that signal. The handset receives the radio signal from the base, converts it to an electrical signal and sends that signal to the speaker, where it is converted into the sound you hear. When you talk to handset microphone, the handset transmits the audio in the same way to base that then sends that audio to telephone line. The base and handset operate on a frequency pair that allows you to talk and listen at the same time, called duplex frequency. There are many generations of cordless telephones in use. Cordless phones first appeared around 1980 in USA. Those earliest cordless phones operated at a frequency of 27 MHz. In 1986, the Federal Communications Commission (FCC) granted the frequency range of 47-49 MHz for cordless phones, which improved their interference problem and reduced the power needed to run them. However, the phones still had a limited range and poor sound quality. In 1994, digital cordless phones in the 900 MHz frequency range were introduced. Digital signals allowed the phones to be more secure and decreased eavesdropping (is was pretty easy to eavesdrop on analog cordless phone conversations). In 1998, the FCC opened up the 2.4 GHz range for cordless phone use. Other countries have also cordless phone systems that operate at different frequency ranges. For example in Europe there has been systems like CT1, CT2 and DECT.CT1 is a simple analogue system that operates at 47 MHz band (8 channels). Also versions for 31 MHz and 900 MHz operation exist (more channels here).CT2 is a more modern system that uses digital radio communications at 864.1-868.1 MHz frequency range. CT2, the second generation of cordless phones, uses a digital speech path in any one of the forty 100KHz wide RF channels in the frequency range 864-868MHz. Instead full duplex operation is obtainedby the use of a digital technique known as Time Division Duplex (TDD). With TDDthe two halves of a telephone conversation are first converted into digital formand then they are divided into a number of small data packets. Each packet isthen compressed to one half of its original size before the two sets of data are interleaved on the same carrier frequency. The CT2 specification defines a Common Air Interface (CAI), which means that all CT2 handsets and base stations can communicate with each other, regardless ofmanufacturer. The modulation methid enployed is two-level FSK with frequencydeviations of (a) 14.4 to 25.2 kHz above the carrier frequency representingbinary 1 (b) 14.4 to 25.2 kHz below the carrier frequency representing binary 0. A single RF channel is used for both directions of transmission using the'ping-pong' version of TDD. Speech signals in either direction of transmissionare sampled and coded into digital form at 32kbits/s. The 2ms duration samplesare transmitted at 72kbits/s in 1ms bursts to allow the bits to be compressedinto packets of data of 1ms duration. Forty RF channels are available do that CT2 is a combined FDMA/TDD system. DECT is a digital telecommunication system standardized by ETSI. It operates at 1880-1900 MHz frequency (uses TDMA modulation). DECT supports 20-500 meters range with both voice and data communications (nowadays the main use is voice). The Digital European Cordless Telephone system uses a cellular radio-like technologyThe DECT system uses a three dimension cellular layout in which there may be cellsabove and below one another as well as side by side layout and is designed for high density use. The DECT system uses the frequency band 1.88-19GHz and this band is divided up into ten separate carrier frequencies. In turn, each carrier frequency is divided into 23 time slots, any two of which are used for a conversation The system provides 32kbit/s voice channels using TDD. DECT uses FDMA/TDMA/TDD techniques to provide 120 duplex channels using 10 separate carrier frequncies and multiplexing 12 send channels and 12 receive channels onto each carrier. The bit rate per channel is 1152kbits/s and the modulation is GMSK with a frequency deviation of +/- 228kHz and a carrier spacing of 1728kHz. PHS is a personal communications system, which supports bothprivate use (i.e. for use as cordless telephone or wireless PABX extension) andpublic use (i.e. for use in the public PHS service). It is widely used in Japan.PHS generally operates at 1895-1906.1 MHz which isdesignated for PHS private use in Japan. PHS equipment that is designed towork in the private and public PHS band operate at 1895-1918.1MHz.

    Computer telephony

    Computer can be used to perform lots of fuctions in modern telephony systems. This link collectains mostly information on linking computers and traditional telephone systems. There is a separate link section for Internet telephony. Computer telephony integration (CTI) is a term to which many are becoming accustomed. It encompasses an entire industry, devoted to the closer integration of telephony systems with computer-control devices, as well as an ever-expanding range of applications. At the forefront of this industry are innovative products, built using hardware able to terminate digital telephony tier 1 (T1) and E1 (T1 European equivalent) trunk interfaces, fax and voice processing resources, voice-over?IP (VoIP) technology, and other standard peripheral devices. Typically, these operate in industrialized chassis housings and act as switches, voice-mail servers, automatic call distributors (ACDs), and nearly any other kind of telco equipment imaginable. The CTI revolution has led to a generation of such equipment, upsetting traditional notions of how telephony networks should be built.

      General information

      • Speech-Enabled Interactive Voice Response Systems Tutorial - Serving as a bridge between people and computer databases, interactive voice response systems (IVRs) connect telephone users with the information they need, from anywhere at any time. Most of today's IVR and transaction-processing applications employ a touch-tone or dual-tone multifrequency (DTMF) user interface. However, applications that allow callers to use their own voice rather than DTMF inputs to complete transactions are rapidly emerging as the latest innovation in telephony-based remote self-service. This tutorial explores the current state of speech-enabled IVR applications, with emphasis on phonetic speech recognition, features and benefits, and development and deployment strategies.    Rate this link
      • The PBX Goes PC - voice-enabled modems and new software standards provide inroads to computer telephone integration    Rate this link
      • Microsoft Telephony Overview - Telephony integrates computers with communications devices and networks. Under classic telephony, the device was a telephone and the network was the Public Switched Telephone Network (PSTN). Modern telephony continually expands the range of devices and networks, and currently covers devices such as cameras and networks such as the Internet. This document illustrates Microsoft Telephony architecture, and is hyperlinked as a basic roadmap to the material in the TAPI documentation. Please note that TAPI is not limited to PSTN, ISDN, or TCP/IP transport.    Rate this link

      Accessory circuits

      • Circuit eliminates PC echoes - long-distance-telephone services available via the Internet often require the PC user to wear headphones of a headset to prevent echo caused by the microphone's picking up the loudspeaker outputs, this circuit eliminates the echo while using the existing PC microphone and speakers    Rate this link

    Leased lines

    A leased line is provided by a telecom company and provides you with direct connection between either two sites or multiple sites. Unlike a dial-up line the leased line is available at all times, but still can go through exchanges There are many configurations of leased lines available. Typically these will be 2-wire or 4-wire circuits. 2-wire circuits have a similar two-wire interface as normal telephone line. Depending on the line type there can be line current present or not. This two-line interface can be used for communication to one direction or for bidirectional communication (like normal telephone line). Four wire interface has one wire pair for transmitted signal and other for receiving signal. Devices connected to leased line wire pairs are typically terminated with a 600 Ohm impedance. Leased line can generally be used to transmit normal audio signals similar to what normal telephone line can carry (expect same signa level and frequency range limitations as normal telephone line). On 2- and 4-wire leased lines, pairs of modems are used to provide point to point full duplex data communication. These modems will typically use the V.22bis, V.32bis or V.34 modulation standards to provide connections For quite short distances there is sometimes plain copper leased line connections available. Those are just plain copper wires wired from one place to another with no active electronics in between. Those can generally carry wider bandwidth than normal telephone lines andcan be used for current loop communications, wider bandwidth audio communications (even broadcast audio) and for high speed data.

    Signaling and coding

    Technical characteristics of tones for the telephone service are listed in ITU-T Recommendation Q.35 (1988). Unfortunately that document is not freely available (can be ordered from ITU-T if you are willing to pay), so the information available in the following documents might be useful if you are looking for free information on telephone line signaling.

      Basic signaling

      When telephone is on-hook, one telephone line wire is quite near to the ground potential and other one carries -48V. When telephone is put off-hook the voltage beween wires going to telephone drops down to the 3 to 9 volt range and typically a current of 20-60 mA will flow through the telephone. The telephone central looksif there is current flowing on the line to get information if telephone is in on-hook or off-hook state.

      To ring your telephone, the telephone company momentarily applies a 90 VRMS 20 Hz AC signal to the line (the voltages and frequencies can vary somewhat country to country).

      For dialling the telephone number the subscriber can use two methods: pulse dialling and tone dialling.

      Pulse dialing or loop disconnect dialing, is pulsing in which a direct-current pulse train is produced by interrupting a steady signal according to a fixed or formatted code for each digit and at a standard pulse repetition rate. The pulses are generated through the making and breaking of the telephone connection (akin to flicking a light switch on and off); the audible clicks are a side effect of this. Each digit in the number is represented by a different number of rapid clicks. In most countries one click is used for the digit 1, two clicks for 2, and so on, with ten clicks for the digit 0. (Two exceptions to this are New Zealand, with ten clicks for 0, nine clicks for 1, and so on, and Sweden, with one click for 0, two clicks for 1, and so on.) Individual digits in a phone number need to be separated with a short pause so as not to bleed into each other. The typical pulse rate for pulse dialling is 10 pulses per second or somewhat less.

      Most phones now use dual tone multi frequency (DTMF, also called touch tone or tone dialing) rather than pulse dialing, but most telephone equipment retains support for pulse dialing for backward compatibility.

      The telephone cental can send various tones to the subscriber to tell the state of the telephone line (for example dialling tone, ringing tone, busy tone etc..).

      DTMF dialling

      Touch-tone dialing, also know as DTMF dialling, is a method of sending signals from telephone customer's premises to central offices and beyond. The idea of touch-tone dialing was first introduced in 1964. Today, most of the telephones in the in developed countries use touch-tone dialling. The advantage of touch tone signaling is that the signaling is voice band signal and the dialing can be done faster than with pulse dialing. The fact that DTMF signaling energy is in the voice frequency band, makes it possible to transmit signaling information (12 distinct signals) to any point in the telephone network to which voice can be transmitted. This makes it possible to use DTMF signals for remote control functions in additions to normal dialling. DTMF tones are used for controlling many modern automated telephone answering services.

      DTMF (Dual-tone Multi Frequency) is a tone composed of two sine waves of given frequencies. Individual frequencies are chosen so that it is quite easy to design frequency filters, and so that they can easily pass through telephone lines (where the maximum guaranteed bandwith extends from about 300 Hz to 3.5 kHz).

      1209 Hz

      1336 Hz

      1477 Hz

      1633 Hz

      697 Hz





      770 Hz





      852 Hz





      941 Hz





      The tones are generated on normal phones based on the keys you press. This table resembles a matrix keyboard. The X and Y coordinates of each code give the two frequencies that the code is composed of. Notice that there are 16 codes; however, common DTMF dialers use only 12 of them. The "A" through "D" are "system" codes. Most end users won't need any of those (are used by some PBX systems for special functions). In PSTN applications dedicated telephony circuits are used to generate DTMF (for example, MT8880). Also microcontrollers are used to generate those tones with help od suitable software (either sampled sound playback or software generating right tones with suitable algorithm).

      In computer modems and GSM modules the sending of DTMF tones are controlled by AT commands. Just send one AT command AT+VTS=X and generate correct tones via software. AT+VTS=1 generates the DTMF tone 1 for 100 ms.

      Often, dedicated integrated circuits or DSP software are used to detect DTMF signals. It is not easy to detect and recognize DTMF with satisfactory precision. The MT8870 is a commonly used complete DTMF receiver integrating both the bandsplit filter and digital decoder functions.


      • Ensuring Voice Quality with Adequate Tail Length - Designers who worked on the POTS never dreamed of the challenges facing those working on next generation telecommunications equipment. Today's engineers live in the digital world where equipment designs are required to codify, compress, cancel echo, control jitter and loss, packetize, switch, route, and bill, and do it all more quickly and efficiently than their competitors. It is easy to lose sight of the need to provide a quality voice connection when the principal design goal is to maximize the volume of data the device can handle. Given these concerns, worrying about the length of the circuit tail delay in an echo-cancellation algorithm seems downright old fashioned and low on the design requirements list. But failure to appreciate tail delay will adversely affect voice quality.    Rate this link
      • E-Series Recommendations Excerpts    Rate this link
      • How message waiting lights work - description of different message waiting light systems    Rate this link
      • A Brief Introduction to CCS7 - CCS7 is somewhat analogous to TCP/IP in that it is a protocol that allows networked computers (in this case telephone switches) to talk to each other.    Rate this link

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