viernes, 30 de julio de 2010


The word "Telecommunications" is nothing but a term which defines devices and systems which is used to transmit electronic or optical signals across long distances.It is only because of this telecommunication, people around this world are able to contact one another,and they are able to access the information at an instant,and they are also able to communicate from remote areas.


The process of telecommunications usually involves a sender, who transmits the information and one or more recipients, and both of them are linked by a technology.Usually this technology which is used to connect thesender and a receiver includes a telephone system,for example,which transmits the reqired information from one place to another.

This process of transfering message is not only confined to a very small are but telecommunications enables people to send and receive personal messages across town, between countries, and to and from outer space. It also provides the key medium for delivering the news, datsa, information, and also for entertainment purpose,such as broadcasting live sports programmes.

Usually the telecommunications devices convert different types of information, such as sound and video, into electronic or optical signals.The reason why we use electronic signals for transmission is that it typically travel along a medium such as copper wire or are carried over the air as radio waves.

Apart from this, Optical signals are specially used at present as they typically travel along a medium such as strands of glass fibers or optical fibers.Normally when a signal reaches its destination, there will be a device on the receiving end, and this converts the signal back into an understandable message, such as sound over a telephone(where electrical is converted into sound), moving images on a television, or words and pictures on a computer screen.

For the transmission of the telecommunication message, there are a variety of ways used.The number of ways of sending a message and it also depends on number of sender and receiver,for example

1)One sender to a single receiver - point-to-point

2)One sender to many receivers - point-to-multipoint


In the case of the personal communications, such as a telephone conversation between two people or a facsimile (fax) message (see FacsimileTransmission),point-to-point type of transmission is used  transmission.

For broadcasting purposes we generally use Point-to-multipoint telecommunications, and this type provides the basis for commercial radio and television programming.


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Telephone-Line Basics For Modems

Modem connections to the telephone service are made using two wires (ring and tip) that are used in a standard telephone jack. The wires are named for the plug wires used in the original telephone lines by which telephone operators would manually connect two telephones at the phone company switchboard. There are two versions of the telephone jack:
  • Half-duplex: The RJ-11 has only two wires, which make up one line. Therefore, only one signal can be sent or received at a time.
  • Full-duplex: The RJ-12 uses four wires to make up two lines; it can be used to simultaneously send and receive.

Multifunction Modems

Most modems offer some form of fax capability, along with software that adds functions beyond the average, small, stand-alone fax machine. Such a modem is usually labeled a fax/modem. They can store faxes, both incoming and outgoing, for reference or online reading. Most allow direct faxing of a document from a word processor, generally by using the print command to send the pages to the modem, where they are converted on the fly to the bitmap form used to send and receive fax transmissions. Many programs let you to automatically attach a predesigned cover sheet with each fax.
Another addition to the basic data out/data in modem is voice mail. Here, the PC and telephone work just like an answering machine. If the phone rings and the modem does not detect either a data or fax tone, it switches modes and streams a recorded message (the outgoing message). The caller can be prompted to record a message for the owner, and in some cases the modem will even forward a pager call or fax with the message contents.

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jueves, 29 de julio de 2010


IP telephony testbed

Within Mitretek Systems' Advanced Telecommunications Laboratory (ATL), a VoIP testbed has been set up. The testbed consists of two PC clients, two IP gateways, four analog phones, and a PC-based private branch exchange (PBX) (see Figure 1). The purpose of the testbed is to measure and calibrate traffic generated from the voice conversations on the PCs and the analog phones. To collect this information, a packet sniffer, Observer from Network Instruments, has also been installed in the VoIP testbed.

Figure 1: IP telephony testbed connectivity
The IP gateways use the standard H.323 call control protocol and several standard codecs: GSM120, G.711, and G.723.1. The PC-based PBX served as a voice switch in the testbed. The PC clients were connected to a hub with private IP address space. Different codecs were used in measuring the packet delay, jitter, packet loss, and bandwidth utilization.
Both PC-to-PC and phone-to-phone testing was conducted in an intranet environment. In the phone-to-phone experiment, the end-to-end delay was measured as 274 ms. A large portion of the delay is due to the analog/digital conversion, compression, packetization, and OS overhead in the IP gateways. When faster digital signal processors (DSPs) become available, the delay introduced by the equipment should be significantly reduced. The purpose of conducting intranet testing is to provide a performance baseline. When this experiment is expanded to the Internet, the additional delay elevated by Internet can be better understood.
Table 1: Packet size distribution over 30 seconds

Packet Size (Bytes)
<=64 65-84 85-128 129-512 513-1024 >1024
6.4 kbps
Listener 0.20% 81.20% 0.70% 0.30% 0.00% 17.50%
Talker 0.10% 98.30% 0.90% 0.70% 0.00% 0.00%
5.333 kbps
Listener 0.00% 81.40% 1.10% 0.30% 0.00% 17.20%
Talker 0.10% 98.10% 1.30% 0.50% 0.00% 0.00%
64 kbps
Listener 2.30% 0.30% 1.90% 50.60% 0.00% 44.80%
Talker 4.20% 0.60% 3.50% 91.70% 0.00% 0.00%

Table 1 represents the results from a 30-second half-duplex voice session. The columns are packet sizes. With the G.723.1 coder, 81.2% of the packets from the Listener were 64 to 84 bytes in size. When using G.711, the talker sends data out in 310 frames. The remaining packets from the talker to listener are called control packets from the application ranging in size from 64 to 128 bytes. When using the G.723 codecs, the data are in 78-byte frames. When the listener sends data to the talker, the packet size is between 84 and 100 bytes, regardless of the codec. Other packets are from the Observer software or are call control packets from NetMeeting.
Additional tests will be taken in a mixed voice and data environment. The data behavior is highly application dependent. Instead of using a generic data generator to produce the necessary data load, a script language, SILK by Sagueway, which emulates the application transactions, will be programmed to produce Microsoft Word, Microsoft Excel, Microsoft PowerPoint, and Web traffic. It is felt that combinations of these application transactions represent the actual network data behavior better than the traffic generated by the network management device. However, the emulation approach is appropriate only for a reasonable system size. For a large-network system, simulation and analytic models are required to facilitate the analysis.

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Anatomy of a Cordless Telephone

To illustrate the parts of a cordless telephone, we will show you the inside of this one from General Electric (GE). It was made in 1993 and operated in the 43-50 MHz range.

GE cordless phone, including handset and base unit

As mentioned above, all cordless phones have a base and a handset. Let's look at these parts individually.
The base unit of the cordless phone is plugged into the telephone jack on your wall.

Base unit components
If you open up the base and expose the circuit board, you see several components that carry out the functions of the base:
  • phone line interface - receives and sends telephone signals through the phone line
  • radio
    • amplifies signals to and from phone-line interface, user controls and speaker phone (if present)
    • broadcasts and receives radio signals to and from the handset
  • power - supplies low voltage power to the circuits and recharges the battery of the handset

Circuit board in the base of the GE cordless phone
Phone Line Interface
Phone line interface components do two things. First, they send the ringer signal to the bell (if it's on the base) or to the radio components for broadcast to the handset. This lets you know that you have an incoming call. Second, they receive and send small changes in the phone line's electrical current to and from the radio components of the base. When you talk, you cause small changes in the electrical current of the phone line, and these changes get sent to your caller. The same happens when the caller talks to you.
Radio Components
The radio components receive the electrical signals from the phone line interface and user controls (keypads, buttons). The radio components convert the signals to radio waves and broadcast them via the antenna. Radio components use quartz crystals to set the radio frequencies for sending and receiving. There are two quartz crystals, one for sending signals and one for receiving signals. Remember that the base and handset operate on a frequency pair that allows you to talk and listen at the same time (duplex). The radio components include an audio amplifier that increases the strength of the incoming electrical signals.
Power Components
A DC power cube transformer supplies the low voltage required by the electrical components on the circuit board. The power components on the circuit board work with the power cube to supply electrical current to re-charge the battery of the handset.
In addition to the above components, some bases also have audio amplifiers to drive speakers for speaker phone features, keypads for dialing, liquid crystal displays (LCDs) for caller ID, light-emitting diodes (LEDs) for power/charging indicators, and solid state memory for answering machine or call-back features.

You can carry the handset with you throughout the house or outside within the range of the base transmitter. The handset has all of the equipment of a standard telephone (speaker, microphone, dialing keypad), plus the equipment of an FM radio transmitter/receiver.
When you open up the handset, you see these components:
  • speaker - converts electrical signals into the sound that you hear
  • microphone - picks up your voice and changes it to electrical signals
  • keypad - input for dialing
  • buzzer or ringer - lets you know that you have an incoming call
  • radio components
    • amplify electrical signals to and from microphone and speakers
    • send and receive FM radio frequencies
  • LCD or LED displays - indicator lights
  • re-chargeable battery - supplies electrical power to handset

Parts of the GE cordless phone's handset, showing the fronts of the circuit boards

Parts of the GE cordless phone's handset, showing the backs of the circuit boards, the speaker, microphone, ringer and battery
The speaker receives the electrical signals from the audio amplifier in the radio components and converts them into sound. When you remove the cover from the speaker, you see a large round permanent magnet with a hole in the middle and a deep groove surrounding the hole. Within this deep groove is a coil of fine copper wire that is attached to a thin plastic membrane. The plastic membrane covers the magnet and coil.

Close-up view of the speaker in the GE cordless telephone handset

Close-up of the speaker with the top removed

Close-up of the speaker with the plastic membrane and attached coil lifted out. The large metal disc is the magnet.

Close-up of the speaker's plastic membrane with attached wire coil
To hear sounds, the following events happen:
  1. Electrical signals come from the radio components.
  2. The electrical signals travel in the coil of copper wire.
  3. The electrical signals induce magnetic currents in the coil of wire, thereby making it an electromagnet.
  4. The electromagnetic coil moves in and out of the groove within the permanent magnet.
  5. The coil moves the attached plastic membrane in and out at the same frequencies as the changes in electric currents.
  6. The movements of the membrane move air at the same frequencies, thereby creating sound waves that you can hear.
The microphone changes the sound waves from your voice into electrical signals that are sent to the audio amplifier of the radio components. A microphone is essentially a speaker that works in reverse. When sound waves from your voice move the membrane, they make tiny electric currents either by moving a coil of wire within a magnet or by compressing the membrane against carbon dust (see How do microphones work? for details).

Close-up of handset's keypad circuit board with attached microphone and buzzer
The keypad allows you to dial a number. It transfers the pressure from your fingertip on the appropriate key into an electrical signal that it sends to the radio components. Below the rubber keypad is a circuit board with black conductive material under each button (shown above). The keypad works like a remote control. When you press a button, it makes a contact with the black material and changes its electrical conductance. The conductance sends an electrical signal to the radio components indicating that you have selected that number.
Buzzer or Ringer
When the radio components of the handset receive the ringer signal from the base, they send electrical signals to the buzzer. The buzzer changes those electrical signals into sound much like the speaker does. You hear the buzzer sound and know that someone is calling you. In some phones, the speaker is used to make the ringer sound and there is no need for a separate ringer. 

adio Components
The radio components of the handset are like those of the base -- they convert electrical signals from the microphone into FM radio signals and broadcast them at the same frequency as the receiving crystal of the base unit. The radio components also receive radio signals at the same frequency as the broadcasting crystal from the base, convert them to electrical signals and send them to the speaker and/or buzzer (ringer).
Remember that the base and handset operate on a duplex frequency pair that allows you to talk and listen at the same time.
LCD or LED Displays
Most handsets have one or more light-emitting diodes (LED) that indicate various things, such as when the phone has an open line or when the battery is low.

LED indicator light on the handset of the GE cordless phone
Some handsets have an LCD that can display numbers for caller ID features, similar to a cell phone. The LCD may be reflective or backlit so that you can see it when the room light is low.
The handset's battery supplies the power for all of the electrical components in the handset. All cordless phone handsets have a rechargeable battery (nickel-cadmium, nickel-metal hydride or lithium). When the battery runs low, an indicator light on the handset usually lights up or flashes. In some phones, a "beeping" sound may also indicate a low battery. You then recharge the battery on the base of the cordless phone.
The GE cordless phone that we dissected was from 1993. Modern cordless phones have the same functions and much of the same hardware. However, many of the electronic circuits that were once achieved with transistors, resistors and capacitors have been replaced with integrated circuits. This advancement allows the handset to be either smaller with the same functions or the same size with more functions.
In summary, a cordless phone is basically a combination of a telephone and an FM radio transmitter/receiver. Because it is a radio transmitter, it broadcasts signals over the open airways rather than specifically between the base and handset.

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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. Long before cell phones became so cheap that anyone could afford one, cordless phones gave everyone the freedom to walk and talk within the privacy of their own homes.Cordless phones have many of the same features as standard telephones, and there are many models available. In this article, we will examine how these cordless telephones work and see why there are so many different types on the market today.

The Basics

A cordless telephone is basically a combination telephone and radio transmitter/receiver (see How Telephones Work and How Radio Works for details on these two technologies). 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, the handset broadcasts your voice through a second FM radio signal back to the base. The base receives your voice signal, converts it to an electrical signal and sends that signal through the phone line to the other party.
The base and handset operate on a frequency pair that allows you to talk and listen at the same time, called duplex frequency.

Diagram showing how the base unit and handset of the cordless phone talk to each other: Each color represents a different frequency.

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a technical expert on Asterisk, but being fully aware of its gathering momentum as an is a free software / open-source software implementation of a telephone private branch exchange (PBX), I decided to hang out with some Asterisk code jocks for the better of an afternoon.
I was much looking forward to a walkthrough of Asterisk, which when implemented, allows multiple attached telephones to make calls to each other, as well as to connect to the Public Switched Telephone Network.
The setting was an Asterisk tutorial presented last Monday at the Open Source Conference in my hometown of Portland. The cerebral and witty Brian Capouch, assistant professor and chair of the Department of Computer Science at Saint Joseph's College in Rensselaer, Indiana, was the MC.
Brian, who is finishing "Inside and Out: Do-it-yourself Open Source Telephony" for Addison-Wesley, walked us through a series of conceptual slides about telephony, IP telephony, and then Asterisk. And as you have probably guessed by now, that is a basic Asterisk schematic at the top of this post.
Now let us look at some of the slides, and how they come together in a code string for a specific Asterisk-enabled application.
This describes the conceptual basics of an Asterisk- configured call.
As far as extensions are concerned, they are often, but not always numeric,and can quantify a prioritization insofar as how the call being placed is handled.
Applications contained within the code for a specific Asterisk scenario provide instructions for behavior to be executed within that scenario. Here's how Brian summarizes this:
Now, let us take a look at how all of this comes together in what is generally referred to as a "Call Flow:"

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El teléfono es uno de los instrumentos de tecnología con mayor permanencia, particularmente en los negocios. Todos los días estas entidades realizan literalmente miles de llamadas cuyo costo es realmente bajo en comparación con el volumen de dinero que se maneja a través de ellas.
Para la mayoría de las compañías, una porción de este costo es evitable, tomando en cuenta que las redes públicas de telefonía poseen un complejo mundo de tarifas y subsidios, que a menudo resultan en situaciones en donde las llamadas salientes forman solo una parte de las llamadas entrantes. Esto hace que las empresas hayan tenido que confiar a la larga con redes privadas.

En el diseño de una red integrada de voz y datos, debe existir una diferencia marcada entre el limite que existe en el diseño de redes de voz y datos, ya que ambas tratan de establecer sesiones terminales entre usuarios, debido a que el concepto de señalización, direccionamiento y enrutamiento de las mismas son similares.
Los cambios en el diseño de redes integradas de voz y datos están en comprender como estos elementos son conciliados en una misma red. El retardo y las variaciones de retardo, implican una reducción en su impacto, es decir estudiar redes de voz sensitivas al retardo y redes con trafico de datos insensibles al mismo.
Un punto de peso para el diseño de redes, esta en que no todo el tráfico de voz es necesariamente sensitivo al retardo. Por ejemplo, el fax y el correo de voz, no tienen restricciones en tiempo real, como las conversaciones de voz. Por lo que añadir servicios de correo de voz y fax puede ser una justificación, para soportar "voz" sobre redes de datos.
Para esto podemos seguir ciertos pasos para el diseño:
Auditoria de la red
Objetivos de la red
Revisión de tecnología y servicios
Guías Técnicas
Planificación de la capacidad
Análisis financiero

La red telefónica básica se creó para permitir las comunicaciones de voz a distancia. En un primer momento (1.876 - 1.890), los enlaces entre los usuarios eran punto a punto, por medio de un par de cobre (en un principio un único hilo, de hierro al principio y después de cobre, con el retorno por tierra) entre cada pareja de usuarios. Esto dio lugar a una topología de red telefónica completamente mallada, tal y como se muestra en la Figura 12.

Figura 12: Conexión mediante una red completamente mallada
Si se hacen las cuentas, esta solución se ve que es claramente inviable. Si se quiere dar servicio a una población de N usuarios, con este modelo completamente mallado, harían falta Nx(N - 1)/2 enlaces. Por esa razón se evolucionó hacia el modelo en el que cada usuario, por medio de un par de cobre se conecta a un punto de interconexión (central local) que le permite la comunicación con el resto.

Figura 13: Conexión mediante una red en estrella
De este modo la red telefónica se puede dividir en dos partes. La estructura de la red telefónica mostrada en la Figura 13: Conexión mediante una red en estrella es la que básicamente hoy se sigue manteniendo. Lo único es que la interconexión entre las centrales se ha estructurado jerárquicamente en varios niveles dando lugar a una red de interconexión. De este modo, la red telefónica básica se puede dividir en dos partes: la red de acceso y la red de interconexión (Figura 14).
Figura 14: Estructura de la red telefónica

El bucle de abonado es el par de cobre que conecta el terminal telefónico del usuario con la central local de la que depende. El bucle de abonado proporciona el medio físico por medio del cuál el usuario accede a la red telefónica y por tanto recibe el servicio telefónico. La red de interconexión es la que hace posible la comunicación entre usuarios ubicados en diferentes áreas de acceso (CSAs).

Redes Telefónicas
La red telefónica es una red de conmutación de circuitos, dada su extensión y complejidad, se puede clasificar en lo que constituye las propias centrales de conmutación, la parte de interconexión que las une y la parte de enlace con los usuarios o abonados. Atendiendo a este criterio se tiene:
· Red de enlacesEstá constituida por los circuitos que unen las centrales entre sí, utilizando medios de transmisión diversos, como cables de pares o fibras ópticas, que son los que proporcionan la vía de comunicación con otro que cuelga de una central distinta a la suya. Si las centrales que se unen son urbanas, la red de interconexión se denomina red de enlaces urbanos, y si no, red de enlaces interurbanos.
· Redes de abonados.Es el conjunto de elementos de conexión entre los equipos de abonado y la central local a la que pertenecen, de tal manera que cada uno de ellos tiene asignado un circuito único (bucle de abonado).

La conmutación telefónica es el proceso mediante el cual se establece y mantiene un circuito de conmutación capaz de permitir el intercambio de información entre dos usuarios cualesquiera. La imposibilidad de mantener conectados a todos los usuarios entre si, con dedicación exclusiva de ciertos medios para su uso, es lo que hace necesario el empleo de un sistema que permita establecer el enlace para la comunicación solamente durante el tiempo que está dure. Los sistemas que consiguen una mayor eficacia son las centrales telefónicas en sus diversas modalidades.
Atendiendo a la distribución geográfica tenemos tres tipos de redes, las llamadas "urbanas" o de corta distancia, las "interurbanas" o de larga distancia y las "internacionales".
Redes urbanas: Dentro de estas se engloban los circuitos de abonados y los enlaces entre centrales locales, para transmisión en banda base o en baja frecuencia. Normalmente están constituidos por pares de conductores, que al agruparse, forman el llamado "cable de pares", que puede contener hasta varios cientos de ellos, configurados en simétricos y en cuadretes, para una menor interferencia de unos sobre otros.
Redes interurbanas: Esta es la encargada de proporcionar los enlaces entre centrales localizadas en diferentes ciudades; ello hace que las distancias sean mayores y se deban utilizar cables de distintas características a los antes mencionados, con menores pérdidas y una respuesta plana que se consigue de dos formas diferentes: una cargando los cables de pares, y otra, empleando otros medios distintos de los cables de pares, tales como el cable coaxial, fibra óptica, radio enlaces, etc.; todos ellos con una mayor capacidad de transmisión y una mayor fiabilidad.
Redes internacionales: para dar curso al tráfico entre diferentes países se necesita de la interconexión entre centrales internacionales, encargadas de encaminar el mismo. Esta se realiza mediante enlaces de alta capacidad (varios miles de circuitos full-duplex) y fiabilidad, constituidos fundamentalmente por enlaces terrestres, submarinos o vía satélite, repartiéndose al menos entre dos de ellos por razones de seguridad. Los canales empleados son de tipo analógico (FDM/Multiplexaje por División de Frecuencia) o digitales (TDM/Multiplexaje por División de Tiempo)
Las centrales de conmutación son los elementos funcionales encargados de proporcionar la selectividad necesaria, de forma automática, para poder establecer el circuito de enlace entre dos usuarios que desean comunicarse. En ellas reside además todo el control y la señalización propios de la red.
Central Local: A éstas se conectan todas las líneas de abonado, de tal forma que mediante un par físico se une un teléfono con la central. También, se llama central urbana.
Red telefónica conmutada o red telefónica básica
Depende de la Compañía Telefónica y es la red utilizada en las comunicaciones orales por teléfono.
Puede conectarse un usuario, por medio del correspondiente módem, a cualquier otro abonado, identificándose ambos por su número de teléfono.
Ventajas: amplia cobertura, nacional e internacional, y su precio en comparación con las redes de uso exclusivo, ya que se factura según la duración de la comunicación al igual que las conferencias telefónicas.
Inconvenientes: es su baja calidad, al ser una red para voz con un ancho de banda inferior a lo deseable. Se utiliza principalmente para comunicaciones esporádicas y de corta duración. Las velocidades de transmisión oscilan de 1200 a 2400 bps.

Red Iberpac
Promovida por empresas (bancarias), depende de Telefónica y su objetivo es: una red nacional especializada en transmisión de datos.
Grandes nodos de concentración situados en algunas capitales.
Alta calidad y utiliza la técnica de conmutación de paquetes.
Está conectada a las redes públicas citadas en los apartados anteriores y asimismo a las grandes redes internacionales de transmisión de datos: Transpac en Francia, Tymney y Telenet en Estados Unidos, Datapac e Infoswitch en Canadá.

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1.-Modulaciones digitales en el bucle de abonado

Para la transmisión a alta velocidad de señales digitales sobre el bucle de abonado se necesitan moduladores/demoduladores (modems) que operen a frecuencias altas (muy superiores a las de la banda vocal). Dado que la atenuación aumenta con la frecuencia, estos modems deberán utilizar transmisores que trabajen con potencias muy superiores a las empleadas por los sistemas de telefonía para compensar la mayor atenuación que introduce el bucle a alta frecuencia.
Pero no basta solo con que el transmisor inyecte más potencia en el par. Un sistema DSL deberá resolver el problema de la separación de señales asociadas a cada sentido de transmisión (en un sistema full-duplex) sobre un único par utilizando modulaciones que permitan obtener la mayor eficiencia espectral (el mayor número posible de bits por Hz). En este apartado se describen las técnicas existentes para resolver estos problemas.

1.1.-Transmisión "Full-duplex" sobre un mismo par

Antes de pasar a explicar las principales modulaciones empleadas para la transmisión de señales a gran velocidad sobre un par metálico, conviene tener presentes las técnicas que se pueden emplear para la transmisión full-duplex (transmisión simultánea en los dos sentidos) sobre un mismo par de cobre. Las soluciones que se usan son FDD ( Frequency Division Duplexing ) y EC ( Echo Cancellation ).
FDD es una técnica de separación de señales mediante la cual se reserva una parte del ancho de banda disponible para la transmisión en cada sentido. Es una técnica relativamente simple que consiste en el uso de portadoras diferentes para cada sentido de la transmisión de forma tal que los espectros de cada una de estas portadoras (moduladas con la información a transmitir) no se solapen entre sí. Esta es la misma técnica empleada en radiodifusión para separar las señales correspondientes a las emisiones de diferentes emisoras.
Con EC los dos sentidos de la transmisión utilizan todo el ancho de banda disponible por lo que sus espectros se solapan. La separación de la información asociada a ambos sentidos se basa en restar a la señal que se recibe una muestra de la señal transmitida con el fin de recuperar la señal enviada por el otro extremo. Esta es una técnica compleja pues no se trata de una simple resta de señales, previamente es preciso estimar cuál es el retardo introducido por el bucle. El receptor recibe la señal transmitida por el transmisor situado al otro lado del bucle, más varios ecos de su propia señal de los que es preciso conocer cuál es el retardo con el que llegan sus propios ecos para restar correctamente las señales. Estos retardos se determina n en la fase de establecimiento del enlace mediante secuencias de entrenamiento con patrones conocidos de antemano.

1.2.- Modulaciones digitales para la transmisión a gran velocidad sobre pares metálicos

En este apartado se presentan las modulaciones digitales de alta velocidad empleadas por los sistemas DSL para transmitir por el par de cobre. Para una mejor comprensión de los fundamentos e implicaciones de las técnicas de modulación que se van a analizar, se inicia este apartado con una breve introducción (siguiendo una perspectiva histórica) de las modulaciones digitales.

1.2.1.-Modulaciones digitales en banda base 

Una de las primeras modulaciones digitales utilizadas fue la modulación PAM ( Pulse Amplitude Modulation ) que consiste en la transmisión de pulsos rectangulares con una tensión proporcional al código (secuencia de N bits) a transmitir. Su característica principal es que, al utilizar pulsos rectangulares, el espectro de las señales transmitidas empieza en 0 Hz, concentrándose la mayor parte de la energía transmitida en las bajas frecuencias tal y como se muestra en la siguiente figura .
Este tipo de modulaciones se dice que operan en banda base y, por su distribución espectral, son incompatibles con servicios que operen en bajas frecuencias, como el servicio telefónico básico (de 300 Hz a 3.400 Hz) y los accesos básicos RDSI (de 0 Hz a 80 KHz).

1.2.2.-Modulaciones digitales en paso banda 

Más adelante aparecieron las modulaciones digitales paso banda. El punto de partida es idéntico: la información a transmitir se envía en forma de pulsos rectangulares de tensión de una duración determinada, y la amplitud del pulso se corresponde con un determinado código. A partir de este punto empiezan las diferencias: los pulsos a transmitir se multiplican (modulan) por una señal sinusoidal (tono o portadora) de una determinada frecuencia (f 0 ). Esta operación hace que el espectro de la señal transmitida se concentre alrededor de la frecuencia portadora tal y como se muestra en la figura
Este tipo de modulaciones permite la utilización de técnicas FDD para separar los sentidos de transmisión en sistemas full-duplex sobre un solo par de cobre: basta con usar portadoras diferentes por cada sentido suficientemente separadas entre sí. Si en este tipo de modulación se elige adecuadamente la frecuencia de la portadora, se consigue que la señal modulada no se solape con otras señales transmitidas sobre el mismo par, en particular con el servicio telefónico básico y los accesos básicos RDSI.

1.2.3.-Modulaciones QAM 

QAM ( Quadrature Amplitude Modulation ) es una modulación paso banda bidimensional en la que el flujo de bits a transmitir se divide en dos nuevos flujos que se denominan componentes en fase y cuadratura. Los bits que integran cada uno de estos dos nuevos flujos se agrupan en símbolos con NF bits para formar los símbolos de la componente en fase y con NC bits para formar la componente en cuadratura.
Los símbolos de la componente en fase modulan en amplitud (AM) a una portadora de frecuencia f 0 (cos(2. p .f 0 .t)), mientras que los símbolos de la componente en cuadratura modulan en amplitud a otra portadora de la misma frecuencia f 0 , pero en contrafase con la anterior. (sen(2. p .f 0 .t)). Al ser una modulación paso banda, si se elige adecuadamente la frecuencia de la portadora, se consigue la compatibilidad de la señal modulada en QAM con las correspondientes a los servicios básicos de telefonía (analógica y RDSI), y los sistemas full-duplex DSL que usen QAM sobre un único par podrán separar las señales transmitidas y recibidas mediante técnicas FDD.
1.2.4.-Modulaciones DMT 

La modulación DMT ( Digital Multi Tone ) es una generalización de la modulación QAM en la que en lugar de tener una única portadora, se emplean N portadoras equi-espaciadas (denominadas subportadoras), cada una de las cuales está modulada en QAM por una parte del flujo total de bits que se han de transmitir.
En este tipo de modulación el flujo de bits a transmitir se divide en N nuevos flujos, cada uno de ellos integrado por un número variable de bits, número que depende de la relación señal/ruido estimada para la subportadora correspondiente durante la fase de inicialización.
A cada uno de los flujos resultantes se les modula en QAM a su correspondiente subportadora. El conjunto de las subportadoras moduladas en QAM se suman y la señal resultante es la que se transmite por el par metálico.
Al igual que la modulación QAM, de la que no es más que una generalización, la modulación DMT es paso banda: si se elige adecuadamente la frecuencia inicial de las subportadoras, se puede conseguir que el espectro de una señal modulada en DMT no se solape con los espectros de señales correspondientes al servicio telefónico básico o al acceso básico RDSI, y que la separación de las señales transmitida y recibida en sistemas full-duplex sobre un único par de cobre pueda realizarse mediante técnicas FDD.
La modulación DMT es la que en la actualidad se esta usando de forma mas generalizada como tecnología básica en los modem ADSL comerciales.

2.-Detección y corrección de errores 

En los sistemas DSL, como en todos los sistemas de transmisión, se producen errores de transmisión que hay que detectar y corregir. En los siguientes apartados se describen las técnicas disponibles para la detección y corrección de errores de manera genérica, y su aplicación particular a los sistemas DSL.

2.1.-Aleatorización (scrambling) 

La aleatorización en sí misma no es realmente una técnica para la detección y corrección de errores, pero ayuda al correcto funcionamiento de los restantes subsistemas. En este sentido se puede considerar a la aleatorización como un mecanismo de prevención de errores en todos los sistemas de transmisión digital, y los de transmisión digital sobre par metálico en particular.
Los sistemas de transmisión se diseñan suponiendo que el flujo de bits que se transmite (secuencias de 1s y 0s) es totalmente aleatorio. Esta situación en la práctica no siempre se da y son relativamente frecuentes las secuencias largas o cadenas de 0s y 1s consecutivos. Para evitar esta situación se aplica a la secuencia de bits que se transmite una serie de operaciones matemáticas que dan como resultado una secuencia de bits donde las cadenas de 0s y 1s consecutivos no superan en ningún caso una longitud máxima .
En el extremo receptor existe un subsistema, el desaleatorizador ( descrambler ) que efectúa las operaciones inversas y permite la recuperación de la secuencia de bits original.

2.2.-Detección y corrección de ráfagas de errores causadas por ruido impulsivo 

Tal y como se ha descrito anteriormente, el ruido impulsivo causa ráfagas de errores que afectan a varios símbolos consecutivos en los sistemas DSL. Para corregir este tipo de errores se emplean dos técnicas complementarias: codificación FEC y entrelazado.

2.2.1.-Codificación FEC (Forward Error Correction) 

La codificación FEC es un sistema de detección y corrección de errores en el que se añade a la información a transmitir (símbolo) otra información adicional ( overhead ) que permite la detección y corrección de errores (siempre y cuando el número de bits erróneos por símbolo no supere un determinado número función del overhead añadido).
Los codificadores FEC más utilizados son los que usan códigos Reed-Solomon que permiten corregir hasta 16 bytes erróneos sobre un total de hasta 255 bytes. Estos codificadores actúan sobre bloques (símbolos) de L bytes y, mediante una serie de operaciones aritméticas, generan a su salida bloques de R bytes (R > L) introduciendo un overhead o redundancia de R-L bytes. El descodificador Reed-Solomon del receptor, recalculando las operaciones aritméticas, es capaz de detectar la existencia de errores si el número de bytes erróneos es menor o igual que R-L, y es capaz de corregir estos errores si la cifra de bytes erróneos es menor que (R-L)/2.

2.2.2.-Entrelazado (Interleaving) 

El entrelazado de símbolos no permite por sí mismo la corrección de ráfagas de errores debidas a ruido impulsivo, pero sí que potencia el efecto de los codificadores Reed-Solomon . El entrelazado de varios símbolos sobre los que ya se han aplicado técnicas FEC, hace que los impulsos de ruido afecten a más símbolos con pero sobre un número menor de bytes en cada uno de los símbolos afectados (distribución espacial de las ráfagas). De esta forma el descodificador Reed-Solomon del receptor será capaz de corregir errores en un número de casos mayor, disminuyendo la tasa de errores de bits (BER) efectiva.
La utilización del entrelazado conlleva un incremento de retardo (latencia) en la transmisión . Debido a esta latencia adicional, el entrelazado no siempre se utiliza en los sistemas DSL, especialmente cuando estos sistemas se destinan a servicios sensibles al retardo como pueden ser los servicios interactivos de voz o vídeo.

2.3.-Detección y corrección de ráfagas de errores causadas por interferencia entre símbolos 

La interferencia entre símbolos (ISI) es la principal perturbación en los sistemas de transmisión digital, como es el caso de los sistemas DSL. La atenuación introducida por el par, y la variación del retardo de transmisión en función de la frecuencia hacen que los pulsos enviados se solapen con los transmitidos inmediatamente antes y después. La ISI supone una disminución de la relación señal/ruido (SNR) a la entrada del receptor de un sistema DSL, y por tanto un aumento de la tasa de errores de bits (BER). Para paliar el efecto de la interferencia entre símbolos y mejorar las prestaciones de los sistemas DSL se emplea la denominada codificación Trellis.
La idea que subyace en esta codificación es la siguiente: dada una constelación de señales recibidas, hay una relación señal/ruido SNR determinada y, por lo tanto, una tasa de error asociada a esa relación señal-ruido. Si se usase una constelación con menos puntos manteniendo constante la potencia transmitida, mejoraría la relación señal-ruido y la tasa de error, pero la velocidad de transmisión bajaría. En las codificaciones Trellis lo que se hace es lo contrario: en lugar de reducir los puntos de la constelación, ésta se amplía (típicamente se duplica el número de símbolos, lo que supone añadir un bit más por símbolo) manteniendo la potencia transmitida. Esta filosofía supone en principio ir en contra de lo que se pretende ya que, a priori, la relación señel-ruido disminuye. Sin embargo la información añadida por los códigos de Trellis se codifica teniendo en cuenta los símbolos anteriores, de tal manera que la distancia real entre símbolos aumente respecto a la que se tenía antes de expandir la constelación. De este modo, la relación señal-ruido aumenta y la tasa de errores de bit disminuye

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Voz sobre IP y Telefonía IP
Voz sobre IP(VoIP, Voice over Internet Protocol) es una tecnología que permite que se realicen transporte la voz usando una conexión IP, o sea digitalmente y en forma paquetizada. La Voz sobre IP ocupa el protocolo IP para transportar la voz sobre la red y hoy en día se convertido en una alternativa para la PSTN actual. Al transportarse la voz sobre redes IP se esta utilizando un medio lógico/físico existente (Internet), esto ultimo permite bajar los costos en transporte de las señales y ha permitido que en el ultimo tiempo se masifiquen servicios de este tipo. La telefonía IP se refiere a los servicios de voz que utilizan VoIP, hay que diferenciar que VoIP es la forma de transportar y telefonía IP es el servicio final, en este último se dan facilidades de valor agregados tales como ring de llamadas , transferencias, conferencias, identificación de llamadas, autenticación, buzones de mensajes, servicios de operadora, etc. En la sección de implementación se vera en forma practica cada servicio.
En los servicios de telefonía PSTN existe protocolos de señalización de llamadas entre los cuales destaca SS7(Sistema de señalización 7), este estándar define el protocolo y los procedimientos mediante los cuales los elementos de la red de telefonía conmutada pública (la PSTN) intercambian información sobre una red digital para efectuar el ruteo, establecimiento y control de llamadas, para el caso de VoIP destacan H.323 y SIP, para nuestro caso de implementación se opta por SIP, ya que es libre de licencias.
Para que la voz pueda ser transportada en forma digital esta debe ser digitalizada, o sea codificada y decodificada, en este punto es donde intervienen los codecs(codificador-decodificador); éstos a través de software y/o hardware cumplen la función de transformar un flujo de datos (stream). Los códecs codifican un flujo o la señal (a menudo para la transmisión, el almacenaje o el cifrado) y recuperan del mismo, todo este proceso tiene también como objetivo reducir la cantidad de información a transmitir, o sea transmitir el mínimo con el fin de ocupar bajas niveles de ancho de banda. En todo el proceso los codecs generan pérdidas de información, cuantas menos pérdidas se generen y a un menor ancho de banda requerido el codec será mejor.
Uno de los principales problemas de transportar la voz sobre IP esta en que se debe hacer en tiempo real, a diferencia de los servicios tipos de Internet tales como Web, Mail, entre otros, por esta razón la mayoría de las implementaciones se realizan sobre UDP/RTCP(User Datagram Protocol /Real time control protocol) y sobre redes con QoS(Calidad de servicio).
Arquitectura del protocolo VoIP
A continuación se muestra un esquema general de la arquitectura del protocolo VoIP, en esta se puede apreciar como y donde actúan los protocolos de señalización y de transporte.
Estructura protocolos VoIP
Ventajas y desventajas de aplicaciones de la voz sobre IP sobre PSTN
La principal ventaja de la integración de servicios telefónicos vía VoIP radica en la independencia de tener que utilizar redes telefónicas actuales de las propias compañías, esto produce que económicamente las soluciones sean muy atractivas, en términos generales podemos encontrar las siguientes ventajas:

  • Costos mas bajos que tecnologías actuales tales como voz sobre TDM, ATM, Frame Relay.
  • Una vez realizada la implementación dentro de su red IP pasa a ser un servicios mas de esta. Las redes IP son un estándar universal para la Internet, Intranets y extranets.
  • Hoy en día existen estándares efectivos (H.323).
  • Interoperabilidad de diversos proveedores, aunque no es universal, ya que existen varios sistemas propietarios.
  • No de depende de tecnologías de transporte (capa 2).
  • Movilidad, ya que vía IP se llamar a cualquier teléfono en cualquier parte del mundo, ideal para empresas con empleado móviles.
  • Reducción de costos en todo tipo de llamadas.
Entre las desventajas presentadas por los servicio VoIP están la calidad de la comunicación (ecos, interferencias, interrupciones, sonidos de fondo, distorsiones de sonido, etc.), que puede mejorar según la conexión a la red(Internet, Intranet). Garantizar la calidad de servicio sobre una red IP, actualmente no es nada simple, los retardos que se presentan en el tránsito de los paquetes y los tiempos de retardos que lleva el procedo de las conversiones (codec) son alteraciones que son de fácil percepción por los usuarios. También el ancho de banda, el cual no siempre está garantizado, va en desmedro del servicio. Otro factor critico y que se debe considerarse en toda implementación es la calidad de servicio (QoS), es clave priorizar el trafico de voz por sobre servicio que no necesiten estar en tiempo real. En general los problemas de la calidad en el servicio telefónico en el protocolo IP van disminuyendo a medida que las tecnologías involucradas van evolucionando. Otras desventajas son:

  • El servicio queda limitado a todos los que estén en conexión directa a la red(Internet, Intranet); algunos servicios no ofrecen la posibilidad de que el computador reciba una llamada y tampoco funcionan vía de un servidor proxy.
  • Exposición a la perdida y el retardo de la información, ya que la información es transportada y dividida en paquetes y una conexión esta compuesta de transmisión de más de un paquete. Estos paquetes pueden perderse, y además no hay una garantía sobre el tiempo que tardarán en llegar de un extremo al otro de la comunicación.
  • No todos los sistemas utilizados por los Proveedores de Servicios de Telefonía por Internet son compatibles (Gateway, Gatekeeper) entre sí. Este ha sido uno de los motivos que ha impedido que la telefonía IP se haya extendido con mayor rapidez. 
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How does a telephone line work?
This question can be answered by examining the diagram in Figure 1 and the following explanation:

Figure 1:  Telephone Line Basics
Figure 1: Telephone Line Basics

There are several different states that the phone line can be in, depending upon what the phone is doing, and what the phone company is doing to the phone. The central office (or nearby equivalent) applies various DC and AC voltages to the two-wire phone loop (labeled "tip" and "ring") during these various stages of call progress. In the idle state, the telephone company central office applies -48(±6) Vdc in series with 500W to 2500W to the "ring" line, and terminates the "tip" line to ground with 0W to 710W. In addition, there is typically up to 1300W of external line resistance between the central office and the "subscriber". When the phone is placed off-hook, the central office goes into dialing mode, applying a dial tone and a DC level of -43 to -79 volts in series with 200W (±50W) on "ring," and terminating "tip" with the same impedance to ground. The same DC voltage and source impedances are present in the talking state (after the connection is made), although the telephone company may, at its discretion, revere the polarity of the DC voltage applied to "ring." Of course, in talking state there are also audio signals superposed on the DC signal, the whole purpose of the telephone! There are two other states. During ring, the phone company applies 89(±2) Vrms at 20Hz, on top of the usual 48Vdc bias. As with the DC signal, the AC ringing signal is applied to the "ring" lead. The official ringing specification is 2 second burst at 6 second intervals. During test mode, the phone company applies various AC and DC test signals to make sure that the network is working properly.

Turning towards an application, a typical modem circuit looks like that shown in Figure 2:

Figure 2:  Typical Modem Circuit
Figure 2: Typical Modem Circuit

The input is a typical common outlet containing leads for the Tip and Ring. The first component to interface with the telephone lie is the Surge Protection device. This device is designed to prevent any damage to "downstream" circuitry in the event of any transients which may occur such as lightning or power crossings. The Ring Detect device acts as a sensor and provides a signal to the modem when a "ring" signal is detected from the central office. The Hook Switch component activates the connection of the Tip and Ring lines. Both the Hook Switch and Ring Detect mechanisms are connected to a control unit.

Surge Protection
This phase of the telephone call is designed to protect the circuitry from any sudden changes in voltage and/or current from the outside line. These changes could result from transients, power crossings, or a lightning strike. According to regulations stated in the FCC's Part 68 rules, all equipment connecting to the U.S. telephone network must meet certain requirements. Of these, there are two main areas of concern: The Metallic Surge Test and The Longitudinal Surge Test. The Metallic Surge Test is defined as the differential surge voltage across the tip and ring conductors, while the Longitudinal Surge Test is defined as the differential surge voltage across the tip and ring shorted together and ground. The purpose of these tests are to simulate lightning and other transients which may occur on telephone and power lines.

The Metallic Surge Test is the application of an 800V peak surge, with a maximum rise-time-to-crest of 10ms and a 560ms minimum decay-time-to-half-crest applied between tip and ring circuits of the EUT (equipment under test). Two surges are applied, one in each polarity, and the EUT must meet the "non-harm" clause included in FCC Part 68. This clause states that it is acceptable if the EUT remains permanently in the on-hook state (line open), but not acceptable for it to fail and remain in the off-hook state (line closed).

The Longitudinal Voltage Surge Test is the application of a 1500V peak surge in each polarity from the tip connection of the circuit to ground, the ring connection of the circuit to ground and from the tip and ring connections shorted together to ground. The surge must have a maximum rise-time-to -crest of 10ms and a minimum decay-time-to-half-crest of 160ms.

Ring Detection Circuitry
The ring detection circuitry detects the ring signal from the central office and indicates the presence of an incoming call. This ring signal is a high voltage AC signal superimposed on the central office DC battery, normally 48V DC. The AC signal is transmitted with a frequency between 15.3 - 68.0Hz and has an RMS voltage of 40 to 150 volts, with a typical ring pattern of 2 seconds on and 4 seconds off. The ring detection circuitry is connected to an input port on the modem or micro controller where it is verified as a valid ring signal.

Hook Switch Circuitry
The on-hook (on-line) and off-hook (off-line) conditions for the phone interface are controlled by the hook switch via a connection with the modem or micro controller host. The hook switch is activated by the modem or host when placing a call or when answering a call in response to an incoming ring signal.

Solid State Technology and the TR115
With the introduction of solid state technology to the telecommunications market, the different phases of a telephone call can now be handled by solid state relays and optically coupled transistors. Solid State Optronics, Inc. has combined the features of a solid state relay and an optically coupled transistor into a single, 8 pin product called the TR115. Figure 3 shows two TR115 devices connected to the telephone lines in a generic application:

In this application, the TR115 devices are used to perform four different functions: Hook Switching, Ring Detection, Loop Current Detection, and Pulse Dialing. For Hook Switch operation the optically coupled relay is used. When a drive current is applied, the relay closes the loop placing the phone or modem in the "Off Hook" mode and prepares it for dialing or receiving an incoming call. Inherent to the optical relay is an isolation voltage of 3750V which provides more than adequate surge protection required by the FCC's Longitudinal Voltage Surge Test. The peak blocking voltage of the relay is 400Vpp, well above voltages typically encountered along the Tip and Ring lines. The LED driving the relay has typical Turn-On currents around 2mA, making it ideal for battery powered applications such as laptop computers where longer battery life is desired.

For Ring Detection and Loop Current Detection, the transistor portion of the TR115 is used. The low turn on levels and fast switching speeds of the photo transistor make it ideal for these functions. In Ring Detection, the central office signal activates the transistor which denotes an incoming call. In Loop Detection, the transistor is used to monitor whether the device is "On Hook" or "Off Hook."

Modern phones use a method known as DTMF (Dual Tone Multi Frequency) to dial a phone number. Earlier phones used a method known as Pulse Dialing. In pulse dialing, the connection to tip and ring lines is interrupted at a high rate to denote various numbers. For instance, when dialing the number 6, the line would be interrupted 6 times. The relay portion of the TR115 can be used for applications where pulse dialing is used.

New Developments
As the telecommunications industry grows, new relay technology is being developed for it. With telecommunication products becoming smaller and more powerful, increased multi-functionality must be designed into tinier packages. Solid State Optronics, Inc. has developed several new products to meet these demands, incorporating multiple functions into low profile, 16 pin SOIC packages.

Pictured below in Figure 4 is a schematic of Solid State Optronics, Inc. latest product designed specifically for telecommunication applications, the STS700:

Figure 4:  STS700 - Multi Function Relay
Figure 4: STS700 - Multi Function Relay

Besides containing a 1 Form A relay and a Photo Transistor, the STS700 has a Darlington Transistor and a Bridge Rectifier incorporated into it. These two added features are included in the package for use in "dry" transformer and optical Data Access Arrangement (DAA) designs.

Figure 5 shows how the STS700 is used in a sample circuit:

The Bridge Rectifier provides the function of current steering to maintain DAA operation and protect the Darlington Bridge during polarity reversals of Tip and Ring wires. The Darlington Bridge, with the addition of a few passive components, functions as an electronic inductor that has the effect of presenting a low resistance to the DC current across the telephone line, and a relatively high impedance for AC signals on the line. For a transformer based design, this enables the designer to use a small coupling transformer (T1) since the telephone loop current is diverted through the Darlington instead of the transformer windings ("dry transformer"). Without the electronic inductor, the loop current would have to flow through the transformer ("wet transformer"), however, since the telephone loop current can be as high as 120mA, the transformer would saturate, causing signal degradation unless the geometry of the transformer becomes much larger. This is especially true for high speed modems such as V.34bis, where return loss must meet or exceed 25dB. Return loss of 25dB is usually not attainable with a wet transformer, and if it is the transformer is too large and expensive for the application.

Solid State Optronics, Inc. is in the process of developing other products based on the 16 pin SOIC package designed for Telecommunications use. These products are all in the STS family, and will be available with the following configurations:
  • Two 1 Form A relays, and a phototransistor
  • Two 1 Form B relays, and a phototransistor
  • 1 Form A relay, 1Form B relay, and a phototransistor
  • 1 Form A relay, and two phototransistors
These devices integrate control of Loop Current Detection, Ring Detection, Caller ID, Hook Switching and other functions all into one package, making them ideal for a variety of telecommunications products.

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have noticed that many people, in particular those who have a telephony background or were working with common IP PBX solutions, try to match their gained dialplan know how, that they acquired over the last couple of years with many, many PBX/IP PBX projects, on OCS 2007. While there are few similarities with other IP PBX systems, there is at least one fundamental difference in the overall dialplan design of OCS 2007:
Traditional PBX and IP PBX dialplans were created mainly from a node-by-node or site-by-site perspective. If you had a node on site A and another one on site B in e.g. a different area code, the design of the dialplan was primarily focused on this particular site with some cross-node/cross-site prefixes in order to allow inter-site phone calls.
Here is an example: We have a PBX on site A with the subscriber number +4969000 with a 4-digit extension range (+4969000-xxxx). On the other hand we have a PBX on site B with subscriber number +4940000 and also 4-digit extension range (+4940000-xxxx). The two PBX systems were connected either physically or virtually with a direct link provided by a TDM (Time Division Multiplexing) carrier so that inter-site calls did not need to use the PSTN.
If now a user on site A with x1212 wants to make a phone call to the user on site B with extension 3699, she/he has to dial a site-prefix (that she/he needs to know of course, imagine this for 100s of sites!) followed by the extension of the user e.g. 88-3699. On the other hand, the user on site B has to dial 87-1212 in order to reach the user on site A with extension 1212.
PBX/IP PBX systems could even be configured to use the cross-site prefixes on incoming PSTN calls. So that a user in e.g. Hamburg could avoid a long-distance call to Frankfurt by dialing the local Hamburg subscriber number (here 000) followed by the cross-site prefix and the extension of the user (here 000-87-1212).
By using cross-site prefixes the dialplan of each site's PBX could be kept unique. At any situation, even though if the same extension e.g. 1212 was given to a user of site A and to a user of site B, the dialplan was unique. If an incoming call from the PSTN on site B wanted to reach extension 1212 it was clear that extension 1212 on site B was meant. If another user on site B just dialed 1212 it was clear that she/he wants to cal extension 1212 on site B and not extension 1212 on site A, because then she/he would have dialed the cross-site prefix.
OCS has datacenter-model oriented architecture! This means it has been designed to provide real-time services (incl. voice/telephony) even for large enterprise environments by setting up a pool of OCS servers located in one or few worldwide datacenters. These datacenters are connected through the company's corporate IP network with one another and all users, independent from their physical location, can connect via IP to these servers in order to receive real-time services. Effectively this means from a telephony perspective that formerly separate telephony nodes will be concentrated in one physical and logical location and therefore the formerly well-known site-oriented perspective on dialplans cannot be kept anymore as the uniqueness of the formerly known extensions cannot be maintained anymore.
So, how do the "extensions" have to be modified in order to become unique within the OCS "worldwide telephony system"?
The PSTN (Public Switched Telephone Network) gives an example of a telephony system where each endpoint has been given a unique numeric address, also called phone number. With this numeric address every endpoint can reach every endpoint and two endpoints do never have the same numeric address. The typical format that we are also all used to from our mobile phones is the E.164 format, usually written with a '+' sign up front. Typical formats are
+<CountryCode><SubscriberNumber> or
OCS uses this format to assign a unique phone number within the worldwide OCS environment to a user. This is one fundamental difference to most other PBX/IP PBX systems.
In the OCS world a phone number will be assigned to a user and not to a device. In the PBX/IP PBX world usually you would say: "This phone over there has extension 156". So everyone initiating a call using this particular phone will obtain the identity that extension 156 represents. If this is usually Bob's extension and Bill is initiating a call from it, callee John will have to assume that Bob is calling him on an incoming call. So Bill has to explain to John at the beginning of the call that he is not Bob. This is fairly easy even for John to notice as Bob and Bill will have a different voice, but in a Unified Communications environment where communication involves all media (e-mail, Instant Messaging (IM), video call, voice call, application sharing, SMS…) an outgoing call from Bill can be answered by John with an IM. Now, John has to be sure that he is communicating with Bob, because otherwise he might reveal information as part of the IM conversation that was not intended for Bill.
In order to avoid such situations in a Unified Communications (UC) environment, two things have to be assured by the system:
1.       phone numbers need to be assigned to users in a worldwide unique fashion
2.       Users need to identify themselves with the system before starting to communicate (e.g. Logon to Windows/Office Communicator using Windows credentials out of Active Directory)
In order to proceed with the example above, unique phone numbers for the sample users in Frankfurt and Hamburg would look like this:
User A1
User B1
User A2
User B2
The phone numbers will be assigned to users in the Active Directory user configuration as part of the communications tab. The "Line URI" field is the place where the user will receive her/his phone number. It's also called msRTCSIPLine attribute. The format has to be in the following way:
tel:+ e.g. tel:+49690001212 or tel:+49400001212

What's now becoming obvious, is that this format is not a really user friendly. Simone in Frankfurt who had extension 1212 with the previous PBX/IP PBX system might argue that it is not very comfortable for her to dial +49690003699 just to reach her coworker Christine who is sharing an office with her and had extension 3699 before the OCS roll-out. Agreed! In order to preserve the users dialing behavior and to allow an easier adoption of the OCS system as an enterprise voice solution, logical "dialing zones" can be built in OCS around a number of users of the same site that have the same dialing behavior. These zones are called "Location Profiles". A Location Profile is a set of regular expressions that modify a called party number to the + format that is uniquely used within the OCS environment to assign phone numbers to users. These regular expressions allow Simone to enter 3699 and based on her Location Profile the number will be expanded by Office Communicator to +49690003699 before sending to the system. This is transparent for Simone and she doesn't need to know this. (After a while Simone will become used to the fact, that it is much easier to call Christine by clicking on a contact in OC or on a button in Outlook, but for the beginning of the adoption she should be able to maintain her previous dialing behavior.) If Simone wants to make an outbound call to the PSTN, she also can keep her behavior to dial "0040…" for a call from Frankfurt to Hamburg (First "0" is an external line prefix and second "0" needs to be dialed within Germany to reach other area codes) as the regular expressions of the Location Profile will convert this called party number to +4940… by replacing the first two leading "0" with "+49".
For the example above we will have two Location Profiles, one for all Frankfurt users named "FRA" and one for all Hamburg users "HAM".
How to create Location Profiles on OCS will be explained in a separate blog article. Location Profiles are assigned to Office Communicator users by Windows Group policy. An article about this topic can be found here. On Office Communicator Phone Edition 2007 users are able to change their Location profiles by themselves.
Up to this point we have seen that local Location Profiles on OCS solve the problem of having duplicate extensions on one worldwide, enterprise-wide voice system as well as the need for cross-site prefixes as known in the site-oriented PBX world.
We have to keep in mind though that endpoints on OCS at the end are not registered with a tel URI (tel:+4940000xxxx) but with a SIP URI (Session Initiation Protocol Uniform Resource Identifier) of a user ( So if Simone in Frankfurt dials 3699 to reach Christine, the regular expressions of the Location Profile will modify this called party number to +49690003699 and after that the User Services application of OCS will try to match this phone number to its entries of the msRTCSIPLine attribute field in the User Database. If the entry matches (this means that the called party has been enabled for Enterprise Voice on OCS), a SIP URI of the called party will be retrieved and a SIP session (SIP session with media voice = phone call) can be established between the two SIP users. If no SIP URI can be found within the OCS User Database that matches the modified called party number, this means that there is no user on the OCS system who has this tel URI (phone number) assigned. This means further that the called party has to be found in the outside OCS world.
To summarize at this point: Here are the three options to establish a voice call using OCS:
1.       To enter the SIP URI (by clicking on contact in Office Communicator, by typing the SIP URI into Office Communicator …)
2.       To enter the tel URI in the + format (like you are used with your cell phone when e.g. dialing international numbers)
3.       To enter a non-E.164 phone number
What happens on OCS?
1.       OCS will send a SIP INVITE message to the entered SIP URI
2.       OCS Translations Application will try to match the tel URI in + format to a SIP URI of his User Database. If successful, SIP URI will be retrieved and SIP INVITE will be sent to the found SIP URI
3.       Office Communicator will apply Regular Expressions of the Location Profile that the user has been assigned to. As a result of this Number Normalization process, a + format number will be generated and sent to OCS. Further on as above in step 2.
If OCS has determined that a call is not in the OCS world (Reverse Number Lookup fails as no SIP URI can be found that matches a dialed + number by a user), the call needs to be handed over to the outside-OCS telephony environment. For voice calls the boundary to the outside world (PBX with Gateway, Gateway connected to PSTN, SIP carrier…) is built by an OCS Mediation Server. While OCS Audio/Video Edge Server builds the boundary to the Internet for calls to Remote users or to other federated enterprises that use OCS as well, is the OCS Mediation Server break in/out point for connections to non-OCS users, currently mostly used for voice calls.
In the example above two OCS Mediation Servers with Media Gateways sitting between the PSTN and the OCS Mediation Servers have been added. One sitting in Frankfurt/Germany and the other one sitting in Hamburg/Germany. After the decision has been made by the OCS system that the endpoint called party number must belong to an endpoint outside of the OCS world, two things have to happen as part of the OCS Outbound Routing application:
1.       It has to be checked by the system whether the user is allowed to dial this number to the OCS-external world.
2.       OCS has to determine the most suitable break-out point (OCS Mediation Server) to hand over this call to the outside OCS world.
In common PBX/IP PBX system class of service rights have usually been assigned to extensions. Extension 1212 was allowed to make internal, local and national calls, while extension 3699 was allowed to make internal, local, national and international calls but no calls to premium numbers (e.g. 900). In OCS users will be tagged with attributes (representing "dialing rights") and not extensions. There can be an attribute called "local" but there can also be an attribute "local_and_national". These attributes are called "Phone Usages" in the OCS world. It is also possible to name the Phone Usages "A" and "B" as while defining Phone Usages they have no immediate relevance on the numbers that a user is allowed to dial. So don't get confused.
One or multiple Phone Usages will be grouped in a so called "Voice Policy". In the screenshot above of the Active Directory user configuration it is shown how a user will be assigned to a Voice Policy. There could for example be a Voice Policy named "InternationalGER-Premium" that includes the Phone Usages "Internal", "Local", "National-Premium", "International".
Phone Usages determine which "Route" a user is allowed to use for an outbound phone call. An OCS Route consists of a regular expression rule, one or multiple Phone Usages and an OCS Mediation Server FQDN (Fully Qualified Domain Name). The user is allowed to make an outbound call using a particular Route when the following two parameters match:
1.       The dialed and normalized + number matches the regular expression rule for this Route
2.       The Phone Usage(s) for a particular route matches the user's Phone Usage attribute (assigned to the user via Voice Policy) for this particular Route.
Here is an example: The Frankfurt user Simone has the Phone Usage attributes:
User Simone:
Location Profile:               FRA
VoIP Policy                          "NationalNoPremiumGER"
Phone Usages:                  "Internal" and "NationalGER-Premium"

OCS Routes:
Regular expression
Phone Usages
Mediation Server FQDN
Internal FRA calls
Internal HAM calls
National GER calls No Premium

She dials 0230001 in order to call for a taxi ("0" is the outbound dialing prefix). With the regular expressions of the Location Profile "FRA" the number will be normalized to +4969230001, Reverse Number Lookup fails as no SIP URI can be matched to this number and a Route needs to be determined to break-out this call to the PSTN. The user Simone has been tagged with two Phone Usages via VoIP Policy "NationalNoPremiumGER". "InternalFRA" and "NationalGER-Premium". The order of these Phone Usages in the VoIP Policy is important! OCS Outbound Routing will first apply regular expressions on the normalized called party number of these routes that have been tagged with the Phone Usage attribute "InternalFRA". The regular expression for an "InternalFRA" route will look like ^\+4969000 as this Route should only be used for calls that are within the Frankfurt company site.
Regular expression
Phone Usages
Mediation Server FQDN
Internal FRA calls
Internal HAM calls
National GER calls No Premium

Since regular expression does not match the called party number, there cannot be a Route found for the Phone Usage attribute "InternalFRA", OCS Outbound Routing will try to find a Route for this number to match for the second Phone Usage attribute "NationalGER-Premium". A Route that matches this Phone Usage attribute would e.g. have the regular expressions ^(?!(\+49(?!(190|900)))) which basically means that the regular expression matches for all phone numbers starting with +49 but not +49190 or +49900, which are premium numbers. So if Simone would have dialed a premium number the regular expression would not have matched and Simone would not have been able to dial this number.
Regular expression
Phone Usages
Mediation Server FQDN
Internal FRA calls
Internal HAM calls
National GER calls No Premium

Simone has dialed +4969230001. Therefore the regular expression for this route will match. Since Phone usage attribute and regular expression match, a SIP INVITE will be sent to the particular OCS Mediation Server FQDN for this particular route and Simone is able to make this phone call.
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