General Information



Internet telephony (called as well Voice over IP - VoIP) refers to familiar telephony services - voice and fax, which are transported via the Internet, rather than via the PSTN or ISDN.

The basic steps involved in an Internet telephone call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet without the need of establishing a circuit switched connection as it is in the case of ISDN and PSTN. This type of connection is called packet switched. Of course at the receiving end the signal needs to be converted back to analog.

Initially the Internet telephony was introduced (in early 1995) as Internet Phone software designed to run on a 486/33-MHz (or higher) personal computer (PC) equipped with a sound card, speakers, microphone, and modem. Calls were to be conducted over a normal Internet connection and QoS was not possible. This was PC -to-PC service that required both PCs to be equipped with the proper software. The idea developed rapidly and as next step so called IP telephony Gateway servers were introduced, which allowed standard telephones to be used. Basically these gateways convert back an forth voice to be carried on IP and PSTN or ISDN networks, i.e. a call can be originated on a standard telephone connected to a PSTN or an ISDN network, then at a gateway switched to an IP network, and then again yet another gateway back to PSTN or ISDN where it can reach another standard telephone.

Quality of Service (QoS)
The main issue related to VoIP is the so-called Quality of Service (QoS) issue. This has to do with the nature of the Internet.

Not using circuit switched connection from one side provides better utilisation of the network resources (circuit switched connections are established at the beginning of the call and freed when the call is finished being occupied even if there is no voice traffic, e.g. when the parties involved are silenced, whereas packet switched networks "open" the connection just long enough to send a small chunk of data, called packet), but from another, it cannot itself guarantee that the voice packet will be delivered in time and in sequence (packets in Internet normally travel over road that cannot be pre defined - for example IP packages carrying portions of a picture you want to download from a web site in Germany to Switzerland may come to you via different paths - some via US, others via Japan) and without losses (loss of some packets introduces gaps or periods of silence in the conversation, leading to a clipped-speech effect).

Delay occurs during the processing of the packets as well.

Another issue is the limitation of the available bandwidth - Internet connections may well be crowded on certain legs of the Internet network if there is no sufficient bandwidth provided. QoS is what aims to guarantee that VoIP provides a circuit switched fixed line quality of speech. Consequently, being based on IP, a VoIP service may be provided on any carrier that can provide IP and QoS, e.g. xDSL, Cable, LAN, DECT, and specially dedicated IP networks.

Making an IP call
There are a number of ways that you might talk to someone using VoIP. What you may need is one or more from the following: a computer, an IP phone, a standard (non-IP) phone, and a network connection that in one or another way can handle IP. If you've got a computer or a telephone, you can use at least one of these methods without buying any new equipment:
• Computer-to-computer - This is certainly the easiest way to use VoIP but hardly the best from quality point of view. All you need is the software (free or very low-cost one is easily available), a microphone, speakers, a sound card and an Internet connection, preferably a fast one like you would get through a cable or DSL modem. Except for your normal monthly ISP fee, there is usually no charge for computer-to-computer calls, no matter the distance.
• Computer-to-telephone - This method allows you to call anyone (who has a phone) from your computer. Like computer-to-computer calling, it requires a software client. In addition it requires a service provider (this may be the one that provides you with the software) who provides you with the access from the Internet to the fixed network, and which may require a small per-minute charge.
• Telephone-to-telephone - Through the use of IP gateways, you can connect directly with any other standard telephone in the world. What you need is a service provider which gateway you must call in using whatever POTS you have and provide the called party number - your service provider will connect you through their IP-based network which via another gateway close to the destination will switch the call possibly to another POTS.

As far as the IP phones are concerned, there are basically two types: one that connects directly to an IP network and the other that connects to a computer and used to replace the computer's microphone and speaker. Whereas the latter is used just to facilitate your VoIP experience when using a PC, the former needs a network connection that utilises the IP protocol. Such IP phones in fact comprise part of the hardware and software that the computers use for Internet telephony - basically they do not utilise modems in the sense computers do. Examples for networks that can be used are business LANs or IP based PBX; Cable or xDSL are the most likely candidates to bring the IP telephony in your home (apart of the normal way of using computer).



The Digital Enhanced Cordless Telecommunications (DECT) technology is standardised by the European Telecommunications Standards Institute ( First terminals implementing DECT were available on the market as early as in 1992. Swissvoice, at that time Ascom Terminals, took very active role into the development of the extensive set of DECT standards and had one of the first working DECT prototypes. Throughout the time Swissvoice employees have been editors of a number of DECT standards and have chaired various DECT standardisation committees.

The DECT technology is a general digital radio access technology. As a radio access technology its basic aim is to provide cordless access to a core network, PSTN, ISDN, IP or LAN.

A DECT communication system comprises of at least 2 terminals: a Fixed Part (e.g. Base station) and a Portable Part (e.g. a Handset). Additionally, a range extender can be added to the system called Wireless Relay Station.

The Fixed part is normally connected with a wire to a public telephone network (e.g. PSTN, ISDN, IP) or may be a part of another terminal (e.g. a PC). A Fixed Part may comprise one or more radio end points and hence cover a larger area and provide more capacity (usually such are the base stations used in large office environment, whereas base stations used in residential environment normally comprise of only one radio end point).

The Portable Part is the terminal used by the end user to access cordlessly the services provided by the external network or application. As with the fixed part, the portable part could be a part from another terminal or device. Portable Parts may communicate with other telephones through connections provided via the DECT base station and the external network, or, with other portable parts via the base station only (known as internal calls and free of charge). Special DECT modes have been developed that allow direct communication between portable parts without the need of a Base station.

How does a DECT system really work
Of course for start what you will need is a pair of one DECT base station and at least one DECT portable part. If you want to talk with the external world you will need as well a public network connection/line, e.g. a PSTN or an ISDN, IP network can do as well too. If you want only internal connections, e.g. between 2 or more DECT portable parts you may skip the external network - you will need the DECT terminals only and the communication between them will be for free.

First you have to connect the Base Station to the telephone line and to a power supply. As soon as your DECT base station is powered up it will measure all the available DECT radio channels and time slot pairs and establish a list with the best few (less interference - good quality). Then the FP will start transmitting short beams (signals) on the time slot found to be of best quality and will, on regular bases, update its list with best communication channels and if necessary may move its beam to another time slot. When could this happen? For example if another DECT base station, your neighbour's one, starts operation in the area - you need not worry, there 120 possible slots available. Each beam carries information about the capabilities of the Base station and will allow a Portable Part to find a Fixed Part. Beams are transmitted every 10 ms (milliseconds 1s = 1000 ms) for a duration of 0.052 ms which make all in total transmission for approximately 7.5 minutes within 24 hours.



ISDN is the abbreviation for Integrated Services Digital Network. In comparison to the PSTN it stands for:

• excellent speech quality;

• multiple telephone calls with one telephone line;

• higher data rates (e.g. for access to Internet); and,

• a comprehensive set of Supplementary Services.


As indicated by the name, ISDN is a digital network contrary to the PSTN, which is an analog network. Digital signalling is what brings along all these nice new features, for example a more sophisticated handling of noise which gives the superior voice quality, or, multiplexing which brings the possibility of 2 independent connections over one copper pair plus a third signalling channel.


Multiple telephone calls with one telephone line

One of the main characteristics that distinguish ISDN from PSTN network is that ISDN is a digital network, which allows time multiplexing to be applied. The consequence is that multiple, in parallel, channels can be provided over one and the same line.


Supplementary Services are most often offered to the user per subscription - some of them may be included in the user basic subscription, for other the user may be required to request explicit enabling and to pay an extra fee. Even without a fee some SS may need a special user intervention before they are to become active.


Multiple Subscriber Number (MSN)

The MSN service allows for the user to posses a number of ISDN telephone numbers. Numbers are allocated by the operator/service provider and may vary throughout regions and type of subscriptions. These telephone numbers can be freely allocated to various users of the line, e.g. in a family household, where there could be one user "father", another "mother", and yet another "daughter/son", different MSN number can be allocated to each of these users, allowing for each user to possess individual number. Modern cordless ISDN phones, e.g. swissvoice DECT phones, allow for these numbers to be allocated to different handsets, therefore allowing for a call to a particular number to be signalled on a particular handset only providing a higher degree of privacy.


Calling Line Identification Presentation (CLIP)

This is one of the most commonly used SS. In some countries, especially outside of <st1:place w:st="on">Europe</st1:place>, this service is known as Caller ID as well. If activated, the service allows for the telephone number of the caller of one incoming call to be displayed on the display of the call-receiving terminal during the ringing phase. Seeing the number of the caller, the user may decide to answer the call or not.


Even the service is activated it there is no guarantee that the calling number will be displayed. Why? In order to provide you with the number your ISDN network needs first to have it and second to be allowed to send it to you. If the caller comes from an external to you network it is possible that his number will not be provided to your network, hence for your network the calling user/number will be unknown. Secondly, an ISDN user may decide to impose restriction to the network and forbid submission of her telephone number to the called terminal - this is another SS service called Calling Line Identification Restriction (CLIR). If you do not want to receive calls that do not show the calling party number there is a SS offered for such cases too - it is called Anonymous Call Rejection (ACR), which you need to activate.


Call forwarding (CF)

CF is a service that allows for an ISDN subscribers to be reached on other than her dedicated phone, i.e. the user may decide incoming calls to be transferred to any external telephone number, for example to her office number during working hours or a mobile telephone during holiday for example. There are different types of CF to choose from and the transfer of the call can be individually programmed for each multiple subscriber number. The following types of call transfer can be programmed:

• Call Forwarding Unconditional (CFU) - All incoming calls are transferred immediately, regardless of whether there may be someone available to answer the call.

• Call Forwarding Busy (CFB) - Incoming calls are transferred only when the user line is busy, i.e. both B-channels are occupied.

• Call Forwarding No Reply (CFNR) - Incoming calls are transferred only if the call is not answered within several seconds when the user still has the opportunity to accept the call.


Connecting line identification presentation (COLP)

If a number you are calling has been forwarded by its owner and you would like to see the number to which you are forwarded, the COLP service is for this. By activating the "Connected line identification presentation" on your ISDN telephone you can avoid having an expensive call if the called fixed line telephone was forwarded to a mobile telephone for example.


Call waiting (CW)

CW is used to show to a user already engaged in a call, that a new incoming call is present. The new presence of call may be announced by a special ringing tone and if CLIP service is possible, the telephone number of the person calling will be shown on the display. Upon indication for CW, the user may decide to reject, accept or transfer the incoming call to another phone or a telephone answering machine for example.



An ISDN user can maintain 2 calls in parallel, this service is called Call Hold (HOLD). Brokering is an SS that allows the user to switch back and forth between two calls when using HOLD. One of the calls is active at a time and the calling parties that are not actually being spoken to, cannot listen in to the conversation being active. Brokering and HOLD are normally used in combination with the CW service.


Park call (PARK)

Consider the user does not have a cordless ISDN phone but a corded one only and is engaged in conversation through his telephone in one room being connected to the wall socket in that room. Let us assume now that the user wants to move during the call into another room to verify something and this room has as well a telephone wall socket. The PARK is SS handling this scenario. The user can park the existing connection for few minutes, unplug the telephone from the current room, move to the desirable room, plug in the telephone and continue with the conversation after un-parking the call.


Conference calling (3PTY)

A call is usually communication between two sides, if one would like to talk to 2 other sides, i.e. 3 users talking/listening one with/to each other, the 3PTY (which stands for 3 party) service is to be used. 3PTY allows for the user, after one call has been activated, to set up a second call and subsequently activate the "Conference call" service. In this way all three parties can talk to / listen each other at the same time (unlike the case of the HOLD when one of the parties does not here the other two when they speak one to another).


Completion of calls to busy subscriber (CCBS)

Remember the frustrating experience when you are trying to call a number that is busy? The CCBS is for this. If the dialled number is busy, the automatic "Completion of calls to busy subscriber" can be activated. With the automatic completion of calls activated the user does not need to try calling again and finding that the called number is still busy - instead, as soon as busy number finishes the call and goes on-hook, the network will automatically call it and indicate this to you by a special ringing tone.


User-to-User Signalling (UUS) and SMS

If the telephone network supports it, ISDN offers the possibility to exchange short written messages between ISDN telephones. A prerequisite for sending and receiving these mails is that both parties have ISDN telephones that support this service.


Advice of charge (AOC)

AOC provides the user with the charging information being displayed during a call, hence you can control your expenses and complete a call when it is becoming too long and too expensive.



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 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...

How a Plain Old Telephone (POT) works

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
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.

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".



The DSL technology has been in the minds of Telecommunication Services providers since more than a decade (it must have been born sometime around 1985). Initially it was thought as the technology, which will allow the Telecom, companies to deliver video to their customers. This deployments did not really happen and the technology needed to wait another few years when the demand for high speed internet access brought her back to life.
How does it work
Basically, DSL is a broadband communication technology that provides for high-speed access to the Internet and remote networks (e.g. the network in your office) using the phone lines that are already present in your home. The magic is in utilising frequencies that are out of the frequency band dedicated to the plain old telephone service (POTS) -- PSTN or ISDN, as normal voice and fax services are called. On the figure bellow the separation of technologies into the frequency domain is depicted.

In order to utilise the frequencies above the voice audio spectrum, and provide both POTS and DSL services on the same phone lines, special equipment must be installed on both ends (i.e. in your home and in the Telecom operator Central Office (CO)).

The device usually installed in your home is called a splitter. It is used to separate the frequencies of the telephone audio spectrum from the frequencies of the xDSL signals, that is to allow you to use the line for your telephone as before when using it to access the Internet.

As already said a splitter is normally required at both the customer premises and at the far end (CO). However, whether a splitter is required or not in your home depends on the xDSL service being provided, for example the special version of the DSL called G.lite does not require splitter, if DSL is offered over an analog line splitter is not needed either. A splitter-less scenario is shown on the figure bellow.

Currently, the most popular DSL form in use is the Asymmetric Digital Subscriber Line (ADSL). The basic characteristic of ADSL is the difference between the upstream and downstream bandwidth, hence asymmetric, or uneven. In practice the bandwidth from the provider to the user (downstream) will be the higher speed path. This is in part due to the limitation of the telephone cabling system (e.g. the so called near-end crosstalk and attenuation - have you ever been on the phone and heard some other conversation, not yours, in the background, so, that is a crosstalk; as the attenuation is regarded to the fact that when an analog or digital signal traverses across a medium, it fades and this may lead to the inability to recover the signal on the far end.). Other reason for asymmetry is the desire to accommodate the typical Internet usage pattern where the majority of data is being sent to the user (programs, graphics, sounds and video) with minimal upload capacity required (keystrokes and mouse clicks).

What can you use DSL for
At first, it does not require any change to your phoning and faxing habits - you can still use your existing telephone. The news is that now you can talk on the phone and use the Internet at the same time on a single phone line.

Second, your connection to the Internet is much faster. What can you use fast connection for? Here some ideas:
• Faster downloads of anything digital
• CD-quality audio
• Graphics-rich web-sites
• Faster and better multimedia
• High-speed multi-player games
• Watch on-line movies

And finally, your connection to the Internet is always on. What does this mean? You do not need to dial up your internet provider, no more need to logging on and off; no more busy signals; no more waiting… just open your browser and go! -