PSTN Technolgy

General

The Public Switched Telephone Network (PSTN) is the oldest and largest telecommunications network in existence. PSTN is the telephone network that by default most of the world population is connected to if they have a telephone. In most of the cases a connection to another telephone network, e.g., is done on user request.


The PSTN network is sometimes called an analogue network -- in contrast for example to ISDN and GSM, which are digital networks. The reason is that the signals carried over the copper line are in an "analogue form", that is they have continuously and smoothly varying amplitude or frequency. The human voice and a musical instrument are examples of analog signals - both produce complex variations in frequency and amplitude. The digital signals on the contrary consist of a sequence of discontinuous fragments of the original analogue signal. A digital signal is only a simplified copy of an analogue one, however good enough for our imperfect senses. The frequency band used for the traditional telephony services is in the 300-3,400 Hz interval.


The PSTN history is said to have started in 1876 when Alexander Graham Bell, an American - native of Scotland, while conducting electrical experiments spilled acid on his trousers. His reaction "Mr. Watson, come here, I want you", brought his employee named Thomas A. Watson on the run because the words had been carried into Watson's room by ... electricity and cable (until that time the system had been capable of sending only a ringing tone).


Since then the PSTN network has undergone many changes - we do not need anymore a human operator manually to connect our call to who ever we wanted to talk to (yes, sometime ago this was indeed needed), fax and modems have been added to the normal voice telephones, that round dials our grand mums used for putting in the telephone numbers have been long replaced with touch buttons, and so on and so on... To find out more about the extensive swissvoice PSTN products portfolio go to corded PSTN or cordless PSTN.


How a Plain Old Telephone (POT)

worksOn the picture bellow a PSTN network is represented. Basically it consists of a number of Copper pairs which connect the telephone sockets in our homes to a so called Local Exchange representing an open at the user site network circuit (this part of the network is called as well Local Loop, other term connected to this part is the "Last Mile" that represents the Copper pair from the Local Exchange to the user's premises); number of Local Exchanges then are connected over a Transport (Trunk) network, which today is not necessarily a PSTN network and could for example be an ATM or an IP network.






When the user lifts off the telephone receiver (in the case of a cordless telephone like a DECT telephone this event is replaced by a call from the handset to the base station), the network circuit is closed this notifies the system that the user wants to communicate. The system provides the user with dialling tone back, which is indication that the calling user can now dial the telephone number of the (called) user he wants to communicate with. If the number is already available into the telephone (being for example pre-dialled, or stored in a telephone book), this will be transmitted to the network, otherwise the calling user needs to dial it in. Physically, on the line, the number is provided in one of the two possible ways, namely in a "pulse" or a "Dial Tone Multiple Frequency (DTMF)" mode. "Pulse" is the older, still existing, way of doing this job. The numbers are distinguished by the number of electrical impulses sent over the line. The DTMF mode differentiate the numbers by their "melody", which is represented by a pair of tones per each number sent on the line (in addition to the numbers the characters A, B, C, D, * and # have defined melody too). When the Number arrives into the local exchange the network starts a series of circuits connections based on the meaning of the digits into the number, which at the end result in establishing a circuit between the calling user with the called user terminal -- that is why normally today's telephone networks are called circuit switched networks in contrary to IP(link_to swissvoice IP technology introduction) networks which are packet networks. The called user terminal is provided with a ringing tone, which is sent back to the calling user as well (in the case the cordless telephony, e.g. DECT, it is possible that the activities of the PSTN network are hidden from the Handset - this is called network termination in the Base Station - in this case the Base may provide itself, locally, ringing tones.). What remains is that the called user picks up her/his receiver and starts talking. The call is terminated when both parties involved hang on their receivers.

Multiple analogue calls are carried over the same transmission channel by sending them over different frequency - this is called Frequency Division Multiplexing (FDM).


PSTN Services

Voice and Data are the two basic services supported over PSTN. They have different requirements from the point of vie of the quality of service that is qualified as "acceptable". For example in the case of voice, some degradation of quality is allowed to level when the speech could be still understood and some noise is tolerable; at the same time delay is not tolerable, we do not want to get the words one by one with variable intervals. The case with data is to the contrary - delay is tolerable - for example, we do not mind if we see the Web site we are viewing coming in portions and during a variable interval of time (as far as we do not get bored), however "noise" is not acceptable because even little noise during data transmission would mean a "strange" looking web site (if we could understand that this is indeed a web site at all). This special requirement and the nature of the PSTN network itself have made the PSTN very slow when delivering data. The only way to overcome this is to introduce compression techniques that make the data smaller before it is sent over the PSTN.

The introduction of computer-controlled telephone exchanges during the 70s allowed for operators to create new, completely different subscriber services in the network. Today it is much easier for an operator to create a distinctive image for himself and to increase revenue. Some services are charged for, others are free. Many of the free services help to increase the number of successful calls, which in turn generates revenue. These services are normally known under the name Supplementary Services and Value added services.

Different PSTN operators in different countries have chosen to implement different PSTN services which means that not all possible PSTN services could be find in different countries and even within one and the same country depending on the PSTN operator's choice. The following list shows just few of the services that you may have on offer from your operator:

• Calling line identification presentation (CLIP): Also called "caller ID", this service allows a called party to see the telephone number of an incoming call on a display connected to the telephone line. There are two commonly accepted ways of transmitting the CLIP information - via DTMF or Face-Shift Key (FSK) signalling

• Call forwarding: This service re-routes incoming calls to another number.

• Call-back (completion of call to a busy subscriber): If the called subscriber is busy, the caller can order the call-back service, which means that he is queued for connection to the busy number and when that subscriber gets free the network will connect and notify the caller.

• Call waiting: A special signal is generated during a call in progress to indicate that a third party is trying to reach you.

To access and control those services via the telephone, service codes are used (for example the service code 21 is assigned to "call forwarding unconditional"). The subscriber usually activates
these services by lifting the handset and then pressing a "star-digits-hash" combination. Also, the register button (R) is used, for example, when alternating between calls and for three-party calls. The system usually confirms the execution of a service by sending a tone or a recorded message.

In addition to offering you a pure voice service, a telephone can be an extremely efficient tool for a subscriber who wishes to send a message, make a financial transaction, or just send a short message (SMS) to another telephone (most of the swissvoice PSTN telephones support SMS).

The PSTN treats such services as any other telephone connection: It sets up the connection to a proper destination after the subscriber has dialled the number of the service. When the call is answered, direct contact with the destination, e.g. a computer or a message gateway is established. The keypad telephone's DTMF signalling is then the tool mostly used for transferring information from the subscriber over the established connection.

DTMF, however, is not capable to provide in all cases all the means needed - for example, if a SMS message is to be sent/received it would require a full set of numbers and letters to be available which DTMF does not provide - instead the already mentioned Phase-Shift Key (FSK) signalling is used.

Of course when using a DECT phone and some corded PSTN phones all these may be well hidden from you behind an user friendly menu provided to you by the handset (all swissvoice telephones do this for example). On the handset display you may see the name of the service and what you may need is pressing a single key to activate it - the telephone itself then will identify the "star-digits-hash" combination needed and will send it to the network.


Connecting Digital devices to PSTN

DECT phones, Faxes and Modems are some examples of digital user terminals that can be connected to a PSTN network. In order such devises to operate the digital signal needs to be converted to an analogue signal, or as it is normally called "to be modulated". This is needed as well on the Transport network if it is a digital one, e.g. ISDN, ATM, IP. Different modulation techniques may be used. For many decades, A/D conversion of voice has been performed employing the so-called PCM coding. This is pure amplitude coding that results in a bi-directional connection having a bit rate of 64 kbit/s.In the case of DECT the voice coding makes use of adaptive differential pulse code modulation (ADPCM), which is described in ITU-T Recommendation G.726. ADPCM results in a bit rates of 32 kbit/s with the same information (and thus the same quality) as that resulting from ordinary 64 kbit/s PCM coding.


In the case of Fax, the so called Group 3 fax standard specified in 1980 performs digital scanning - the first fax standards dealt with analog scanning and were designated Group 1 (1968) and Group 2 (1976). The coding standard used is a type of quadrature amplitude modulation (QAM), such as ITU-T V.27ter/V.29, which contains eight phase and two amplitude positions. It provides a maximum bit rate of 9,600 bit/s. However, capacity has increased since Group 3 was introduced. V.17 is a fax standard that provides up to 14.4 kbit/s. The V.34 standard, which was developed for data communication, provides the same capacity. Fallback is always applied, which means that the sending fax starts to transmit using the highest possible transfer rate and then slows down if there are too many errors or if the receiving fax cannot handle the higher transfer rate.

For Modems the number of coding standards in operation is even greater. The main discriminator is the data speed that can be achieved. The International Telecommunication Union (ITU-T) specifies a number of modems for higher speeds (in excess of 48 kbit/s) to be used in the frequency range 60-108 kHz. The standards for these modems are V.36 and V.37. It is also customary to combine modem standards with a compression standard that suits the specific application (fax or file transfer) to further increase available transmission capacity. V.42bis and MNP 5 are two common compression standards.