Sunday, March 30, 2008

Technologies : Common Channel Signaling (SS7)

Common Channel Signaling (SS7)
The signaling system #7 (SS7) is an international standard network signaling protocol that allows common channel (independent) signaling between telephone network elements. SS7 system protocols are optimized for telephone system control connections and they are only directly accessible to telephone network operators.

Common channel signaling (CCS) is a separate signaling system that separates content of telephone calls from the information used to set up the call (signaling information). When call-processing information is separated from the communication channel, it is called “out-of-band” signaling. This signaling method uses one of the channels on a multi-channel network for the control, accounting, and management of traffic on all of the channels of the network.

An SS7 network is composed of service switching points (SSPs), signaling transfer points (STPs), and service control points (SCPs). The SSP gathers the analog signaling information from the local line in the network (end point) and converts the information into an SS7 message. These messages are transferred into the SS7 network to STPs that transfer the packet closer to its destination. When special processing of the message is required (such as rerouting a call to a call forwarding number), the STP routes the message to a SCP. The SCP is a database that can use the incoming message to determine other numbers and features that are associated with this particular call.

In the SS7 protocol, an address, such as customer-dialed digits, does not contain explicit information to enable routing in a signaling network. It then will require the signaling connection control part (SCCP) translation function. This is a process in the SS7 system that uses a routing tables to convert an address (usually a telephone number) into the actual destination address (forwarding telephone number) or into the address of a service control point (database) that contains the customer data needed to process a call.

Intelligence in the network can be distributed to databases and information processing points throughout the network because the network uses common channel signaling A set of service development tools has been developed to allow companies to offer advanced intelligent network (AIN) services.

Figure 1 shows the basic structure of the SS7 control signaling system. This diagram shows that a customer’s telephone is connected to a local switch. The local switch converts the dialed digits to a SS7 signaling message. The SS7 network routes the control packet to its destination using its own STP data packet switches and separate interconnection lines. In some cases, when additional services are provided, SCPs are used to process requests for advanced telephone services. This diagram also shows that the connections used for signaling are different than the voice connections. There are multiple redundant links between switches, switching points, and network databases.


Figure 1: Signaling System 7 (SS7) Network

Thursday, March 27, 2008

Technologies : Public Telephone System Interconnection

Technologies
Some of the key technologies behind the operation of the public telephone network include interconnection lines, network common control signaling, and intelligent call processing. Several types of interconnection systems are used to provide access to different services and systems available through the PSTN. To coordinate the overall operation of the PSTN, a standard common control signaling (CCS) system is used. Intelligent call processing combines these interconnection lines and common control signaling to provide for advanced services such as call forwarding, telephone number portability, and prepaid services.

Public Telephone System Interconnection
There are many types of interconnection options available to connect public telephone systems to other public telephone networks or private telephone networks. The type of connection selected depends on the type of private system, telecommunications regulations, and the needs of the company that uses the private telephone system (e.g., advanced calling features). In addition to standard telephone system connection types, there are also private-line connections that may be used to link private branch exchange PBX systems together.

There are two types of connections used between switching systems: line side and trunk side. Line side connections are an interconnection line between the customer’s equipment and the last switch EO in the telephone network. The line side connection isolates the customer’s equipment from network signaling requirements. Line side connections are usually low capacity (one channel) lines. Trunk side connections are used to interconnect telephone network switching systems to each other. Trunk side connections are usually high capacity lines.

POTS (dial) Line Connections
POTS dial lines are 2-wire, basic line-side connections from an EO with limited signaling capability. Because dial lines are line-side connections, call setup time may be longer than those connections that employ trunk-side supervision.

Direct Inward Dialing (DID) Connections
Direct inward dialing (DID) connections are trunk-side (network side) EO connections. The network signaling on these 2-wire circuits is primarily limited to one-way, incoming service. DID connections employ different supervision and address pulsing signals than dial lines. Typically, DID connections use a form of loop supervision called reverse battery, which is common for one-way, trunk-side connections. Until recently, most DID trunks were equipped with either dial pulse (DP) or dual tone multifrequency (DTMF) address pulsing. While many wireless carriers would have preferred to use multifrequency (MF) address pulsing, a number of LEC’s prohibited the use of MF on DID trunks.

Type 1 Connections
Type 1 connections are trunk-side connections to an EO. The EO uses a trunk-side signaling protocol in conjunction with a feature known as Trunk With Line Treatment (TWLT). This connection was originally described in technical advisory 76 published by AT&T in 1981. This interconnection was developed because dial line and DID connections did not provide enough signaling information to allow the connection of public telephone networks to other types of networks (such as wireless and PBX networks). The switch must be equipped to provide TWLT, or its equivalent to offer Type 1 service. As a result, type 1 is not universally available. The TWLT feature allows the EO to combine some line-side and trunk-side features. For example, while trunk-side signaling protocols are used, the calls are recorded for billing purposes as if they were made by a line-side connection.

Type 1 connections are usually used as 2-way trunks. Two-way trunks are always 4-wire circuits, meaning they have separate transmit and receive paths, and almost always use MF address pulsing and supervision. The address pulsing normally uses wink-start control. One-way Type 1 connections can be provided on a 2-wire basis using E&M supervision or reverse battery like the DID connection.

Integrated Services Digital Network - Basic Rate Interface Connections (ISDN-BRI)
ISDN-BRI connection provides two bearer channels, each using a 64 kbps digital channel, as well as a 16 kbps data link for signaling messages. This 144 kbps combination is referred to as 2B+D, which signifies two bearer channels and one data channel. The bearer channels provide the voice portion while the data channel is used to transfer SS7 signaling messages. EO switches must have an ISDN-BRI interface and software installed to supply this connection.

Integrated Services Digital Network - Primary Rate Interface Connections
Another variation of Type 1 is the Integrated Services Digital Network - Primary Rate Interface (ISDN-PRI). It has capabilities similar to the ISDN-BRI but employs 23 bearer channels and one signaling channel, or a 23B+D configuration. The ISDN-PRI interconnection is provided using a standard DS1-level interface that would normally provide the equivalent of 24 voice channels. It offers the same calling capabilities as noted for the Type 1 and ISDN-BRI connections.

Type 2A Connections
Type 2A connections are true trunk-side connections that employ trunk-side signaling protocols. Typically, they are two-way connections that are 4-wire circuits using E&M supervision with multifrequency (MF) address pulsing. The address pulsing is almost always under wink-start control. Type 2A connections allow the other public or private telephone network switching systems to connect to the PSTN and operate like any other EO.

Type 2A connections may restrict calls to specific NXX (exchange) codes and access to operator services (phone number directories, emergency calls, freephone/toll free) may not be permitted. For some interconnections, additional connections (such as a type 1) may be used to supplement the type 2A connection to allow access to other operator or network services.

Type 2B Connections
Type 2B connections are high usage trunk groups that are used between EOs within the same system. The type 2B connection can be used in conjunction with the Type 2A. When a type 2B is used, the first choice of routing is through a Type 2B with overflow through the type 2A. Because the type 2b connection is used for high usage connections, it can access only valid NXX codes of the EO providing that it is connected to. Type 2B connections are almost always 4-wire, two-way connections that use E&M supervision and multifrequency (MF) address pulsing.

Type 2C Connections
Type 2C connections were developed to allow direct connection to public safety centers (E911) via a tandem or local tandem switch. This interconnection type must provide additional information such as the return phone number (complicated on mobile telephone systems) and the location of the caller. This information is passed on to a public safety answering point (PSAP).

Type 2D Connections
Type 2D interconnection lines allow direct connection from an operator services system (OSS) switch. The OSS switch is a special tandem that contains additional call processing capabilities that enables operator services special directory assistance services. The type 2D connection also forwards the automatic number identification information to allow proper billing records to be created. Type 2D connection will normally use trunks employing E&M signaling with wink start, and multifrequency (MF) address pulsing.

Type S Connections
Type S connections transfer signaling messages that are associated with other interconnection types (out-of-band signaling). The type S is a data link (e.g., 56 kbps) that is used to connect the signaling interfaces between switches. Type S connections permit additional features to be supported by the network such as finding and using call forwarding telephone numbers. Because type S connections cost money, some smaller public telephone networks do not offer or use type S connections.

Figure 1 illustrates some of the different types of private to public telephone system interconnection. This diagram shows some groups of phone lines (e.g., dial line, Type 1) that provide limited signaling information (line-side) that primarily interconnect the PSTN with private telephone systems. Another group of lines (Type 2 series) are used to interconnect switching systems or to connect to advanced services (such as operator services or public safety services). The interconnection lines (trunk-side) provide more signaling information. Also shown is the type S connection that is used exclusively for sending control signaling messages between switching system and the signaling system 7 (SS7) telephone control network.


Figure 1: Private to Public Telephone System Interconnection

Sunday, March 23, 2008

Telecom : Market Growth

Market Growth
By 2001, there were just over one billion telephone lines in the world and the growth for new telephone lines is over 7% per year. The market for public telephone networks is changing from delivery of voice service to data services. In 2000, approximately 97% of all residences in the United States had telephone service. Telephone voice traffic (measured in minutes) has been growing at a rate of nearly 8% per year.

Voice Service
Voice service (telephone line) is any service or feature accessible through the LEC/CLEC or IXC that can be accessed via a standard analog or digital telephone. The key reasons for growth in the number of telephone lines include dial-up Internet access, fax telephone lines, and mobile telecommunications. Figure 1 shows the growth of new telephone lines worldwide. This chart shows that telephone service subscribers continues to grow over 7% each year. Growth had been fairly level at about 50 million additional telephone lines per year. However, during 1999 more than 100 million new telephone lines were added. The recent surge in the number of telephone lines can be explained by the fact that more areas of the world are adding infrastructure to support new technologies that use telephone lines.


Figure 1: Worldwide Telephone Market Growth


Data Transfer
Data transfer is the act of moving data through a network from one data source to another. Generally theses sources are computers and they interface with the network via modems or channel service units (CSU’s). Data transfers can occur over a dial voice grade connection or via a dedicated line.

In 2001, the number of customers that use the Internet was increasing at a rate of nearly 40% a year while data traffic on the Internet (amount of data per user) is expanding at a rate of nearly 100% per year. The amount of data that was transferred over the Internet in the United States in 2000 averaged 27,500 terabytes (1,000 billion bytes) per month. The data transmission on private networks grew 500% between 1997 and 2000 with an average of 3,000 terabytes per month transferred in the United States. Figure 2 shows the data transmission growth within the public telephone networks.


Figure 2: Data Transmission Growth.

Friday, March 21, 2008

PSTN: Numbering Plan

Numbering Plan
A numbering plan is a system that identifies communication points within a communications network through the structured use of numbers. The structure of the numbers is divided to indicate specific regions or groups of users. It is important that all users connected to a telephone network agree on a specific numbering plan to be able to identify and route calls from one point to another.

Telephone numbering plans throughout the world and systems vary dramatically. In some countries, it is possible to dial using 5 digits and others require 10 digits. To uniquely identify every device that is connected to public telephone networks, the Comite Consultatif Internationale de Telegraphique et Telehonique (CCITT) devised a world numbering plan that provides codes for telephone access to each country. These are called country codes. Coupled with the national telephone number assigned to each subscriber in a country, the country code telephone makes that subscribers number unique worldwide. The International Telecommunications Union (ITU) administers the World Numbering Plan standard E.164 publishes any new standards or modifications to existing standards on the Internet.

Each country defines its public telephone network numbering plans. The United States and Canada adopted the North American Numbering Plan (NANP) that allows the two countries to appear as one when dialing internally. Each country has a country code prescribed by the World

Numbering Plan so they are accessed internationally as separate entities. The NANP is based on 10 digit numbering (NXX-NXX-XXXX). The number consists of a 3-digit area code, a 3-digit central office code, and a 4-digit line number. The first three digits (NXX) are the Numbering Plan Area (NPA) or area code. It is this 3-digit code that designates one of the numbering plan areas in the North American Numbering Plan for direct distance dialing. Originally, the format was N0/1X, where N is any digit 2 through 9 and X is any digit. From 1995 on, the acceptable format is NXX.

With the massive requirement for telephone numbers generated by Internet access, fax machines, and cellular telephones, new area codes are being placed in service at an all time high rate. This is causing the telecommunications industry and standards bodies in North America to consider the implementation of “number portability”. When this occurs each subscriber will be assigned telephone numbers permanently (e.g., all subscribers in North America will dial ten digits to make a local call and take their number with them when they move.).

Tuesday, March 18, 2008

PSTN : Switching Systems

Switching Systems
Switching systems are assemblies of equipment that setup, maintain, and disconnect connections between multiple communication lines. Switching systems are often classified by the type of network they are part of (e.g., packet or circuit switched) and the methods that are used to control the switches. The term “switch” is sometimes used as a short name for switching system. Public telephone switching systems have many switches within their network. A typical switch can handle up to 10,000 communication lines each.

Early switches used mechanical levers (crossbars) to interconnect lines. Modern switches use computer systems to dynamically setup, maintain, and disconnect communication paths through one or more switches. True computer-based switching came about through the introduction of the electronic switching systems (ESS’s). ESS EOs did not require a physical connection between incoming and outgoing circuits. Paths between the circuits consisted of temporary memory locations that allowed for the temporary storage of traffic. For an ESS system, a computer controls the assignment, storage, and retrieval of memory locations so that a portion of an incoming line (time slot) could be stored in temporary memory and retrieved for insertion to an outgoing line. This is called a time slot interchange (TSI) memory matrix. The switch control system maps specific time slots on an incoming communication line (e.g., DS3) to specific time slots on an outgoing communication line.

The public telephone network switching system architecture uses a distributed switching system that has a hierarchy of switching levels. Distributed switching systems connect calls through the nearest switching system. With distributed network architecture, the call processing requirements are distributed to multiple points. Using a multilevel hierarchy structure for switching systems allows switching to occur at lower levels of switching unless the telephone call must pass between multiple switches. At that point, the call is passed up to a higher-level switch for transfer to more distant locations.

In conjunction with distributed network architecture, the ability to perform “dynamic routing” furthers the network’s resiliency to faults. Sometimes called “adaptive routing”, dynamic routing automatically re-routes communication paths or circuits as the network traffic levels (e.g., levels of congestion) change or as paths go in or out of service.

A key part of public telephone networks is system reliability. As a result, in the event of equipment failure in such a network, backup (redundant) equipment must provide for continued service. Although this increases the reliability of switching systems, it also increases the system cost (for additional backup equipment) and complexity (recovery management systems).

Public telephone switching systems use EO telephone switches to connect the telephone network to end customers. These switches serve as an end node switch that and provides local dial and access to local and long distance services. Switches that are used to interconnect switches to each other are called tandem switches.

Some systems use mini-switches called remote digital terminals that are located near the EO switch. These mini-switches act as concentrator lines of voice channels between the end customers and the EO switching system. Concentrators grouping multiple communication lines into more efficient trunked (multi-channel) lines.

Wednesday, March 12, 2008

PSTN : Local Loop

Local Loop
The local loop is the connection (wireless or wired) between a customer’s telephone or data equipment and a LEC or other telephone service provider. Traditionally, the local loop (also called “outside plant”) has been composed of copper wires that extend from the EO switch. The EO is the last switching office in the telephone network that connects customers to the telephone network.

The EO switch cables meet the copper (or other types of lines) at the main distribution frame (MDF). The MDF is a wiring rack that allows technicians to splice the local loop lines with the lines from the switching system. Local loop lines leave the MDF in bundles (possibly thousands of wires in each bundle) and arrive in other junction points such as local distribution frames (LDF). The LDF allows the connection of the final connection (the “drop”) to the business or residence. At the entry to the customer’s location, there is often a network termination (NT) device that isolates the telephone network from the wiring inside the customers building.

Figure 1 depicts a traditional local loop distribution system. This diagram shows a central office (CO) building that contains an EO switch. The EO switch is connected to the MDF splice box. The MDF connects the switch to bundles of cables in the “outside plant” distribution network. These bundles of cables periodically are connected to local distribution frames (LDFs). The LDFs allow connection of the final cable (called the “drop”) that connects to the house or building. A NT block isolates the inside wiring from the telephone system. Twisted pair wiring is usually looped through the home or building to provide several telephone connection points, or jacks, so telephones can connect to the telephone system


Figure 2: Local Loop

Tuesday, March 11, 2008

Public Switched Telephone Networks (PSTNs)

Public telephone networks are unrestricted dialing telephone networks that are available for public use to interconnect communication devices. Public telephone networks within countries and regions are a standard integrated system of transmission and switching facilities, signaling processors, and associated operations support systems that allow communication devices to communicate with each other when they operate.

Figure 1 shows a basic overview of the Public Switched Telephone Network (PSTN) as deployed in a typical metropolitan area. PSTN customers connect to the end-office (EO) for telecommunications services. The EO processes the customer service request locally or passes it off to the appropriate end or tandem office. As Different levels of switches interconnect the parts of the PSTN system, lower-level switches are used to connect end-users (telephones) directly to other end-users in a specific geographic area. Higher-level switches are used to interconnect lower level switches.


Figure 1: Public Switched Telephone Network (PSTN)

Switches send control messages to each other through a separate control-signaling network called signaling system number 7 (SS7). The SS7 network is composed of signaling transfer points (STPs) and service control point (SCP) databases. A STP is used to route packets of control

messages through the network. SCP’s are databases that are used by the network to process or reroute calls through the network (such as 800 number toll free call routing). SS7 also provides for the newer features such as incoming call identification and automatic call rerouting used by some service companies that provide 24/7, worldwide dial-in support.

Overview
Public telephone networks include local loops (access lines), switching systems, numbering plans, and are coordinated by network management systems. Post, telephone, and telegraph (PTT) and local exchange carriers (LEC’s) are the established telephone network operators or companies that provide local telecommunications services. For most countries, PTTs are government operated telephone systems. In the United States, LEC’s are granted franchises to provide telephone services to certain geographical areas as mandated by the Federal Communication Commission (FCC). Recently, deregulation and privatization of telecommunication systems worldwide have allowed the creation of a new competing local exchange carriers (CLECs). CLEC’s provide similar services as LEC’s and PTTs. In some cases, CLECs provide services by leasing existing lines from incumbent local exchange carriers (ILECs) and reselling services on these lines. In other cases, CLECs install new communication lines or provide connection by wireless service.

Monday, March 10, 2008

Telecom : Repeaters

Repeaters are devices or circuits that are located between transmitting and receiving devices to improve the quality the signal that is delivered between them. A repeater obtains some or all of the signal from the transmitter, amplifies and may adjust (change a frequency) or filter the signal, and retransmits the signal to the receiver(s).

Repeaters can be analog or digital. Analog repeaters amplify the received signal for retransmission. Analog repeaters amplify both the desired signal and any noise that is added to the communication lines. This limits the maximum number of analog repeaters that can be used and this limits the maximum distance for analog communication lines. Digital repeaters can receive and recreate digital signals. Digital repeaters are also called regenerative repeaters. The regenerative process allows digital signals to be transferred at great distances with minimal errors at the receiving end.

Sunday, March 9, 2008

Channel Multiplexing

Channel multiplexing is a process that divides a single transmission path into several parts that can transfer multiple communication (voice and/or data) channels. Multiplexing may be frequency division (dividing into frequency bands), time division (dividing into time slots), code division (dividing into coded data that randomly overlap), or statistical multiplexing (dynamically assigning portions of channels when activity exists).

When several communications channels are connected over a common channel, a device called a multiplexer is used. The multiplexer combines multiple incoming (input) signals onto one common communications channel through the process of time, frequency, or code sharing. At the other end of the communication line, a demultiplexer device is used to separate the channels (output) at the receiving end.

When a digital channel is divided into multiple digital sub channels, the separate channels are called logical channels. Each logical channel is assigned a portion of the bits from the digital communications channel.


Frequency Division Multiplexing


A device that converts the information signal into a format that is suitable for transmission is called a transmitter. The device that receives and decodes the transmitted signal is called a receiver. When a transmitter and receiver are combined into one device, it is called a transceiver.

Frequency Division Multiplexing (FDM)

Frequency division multiplexing is a process of allowing multiple channels to share a frequency band by dividing up a frequency band into smaller frequency bandwidth channels. Each of these smaller channels provides for a separate communications channel.


Time Division Multiplexing


Figure below shows how a frequency band can be divided into several communication channels. When a device is communicating on a FDM system using a frequency carrier signal, it’s carrier channel is completely occupied by the transmission of the device. For some FDM systems, after it has stopped transmitting, other transceivers may be assigned to that carrier channel frequency. When this process of assigning channels is organized, it is called frequency division multiple access (FDMA). Transceivers in an FDM system typically have the ability to tune to several different carrier channel frequencies.

Carrier signals can co-exist with each other on an FDM system without interference if they are operating at different frequencies. Because the modulating signal slightly changes the carrier signal, this produces small changes in frequency. This results in a single radio signal that occupies a frequency range, depending on the type and amount of information that is changing the electromagnetic wave. The maximum amount of frequency change is typically called the channel bandwidth. Hence, a carrier signal should not typically operate in areas that other radio carrier signals may occupy.

As a carrier signal is modulated (amplitude, frequency, or phase), several other small energy signals at different frequencies are created. Some of the signals produced by the modulation process fall outside the designated frequency bandwidth. Although the amount of energy that falls outside the designated bandwidth is usually small, they may cause interference with other devices that are communicating on other nearby channels.

To help protect from unwanted interference, when multiple carrier signals are operating in an FDM system, a guard band is usually used to protect adjacent carriers from interference. Guard bands are a portion of a resource (frequency or time) that is dedicated to the protection of a communication channel from interference due to radio signal energy or time overlap of signals. While guard bands protect a desired communication channel from interference, the guard band also uses part of the valuable resource (frequency bandwidth or time period) for this protection.

Time Division Multiplexing (TDM)

Time division multiplexing (TDM) is a process of sharing a single carrier channel by dividing the channel into time slots that are shared between simultaneous users of the carrier channel. When a transceiver communicates on a TDM system, it is assigned a specific time position on the carrier channel. By allowing several users to use different time positions (time slots) on a single carrier channel, TDM systems increase their ability to serve multiple users with a limited number of channels by dividing a frequency band into time slots. Time slots are grouped into repetitive frames. Each communication channel is assigned to one (or several) time slot(s) within a frame.

To allow TDM systems to provide continuous voice communication to a transceiver that can only transmit for brief periods, TDM systems use digital signal processing to characterize and compress digital signals into short time-slices. Figure below shows how a single carrier channel is time-sliced into three communication channels. Transceiver number 1 is communicating on time slot number 1 and mobile radio number 2 is communicating on time slot number 3. Each frame on this communication system has three time slots.

Code Division Multiplexing (CDM)
Code division multiplexing uses a method of spreading an information signal using different codes on a wide bandwidth communication channel (typically digital signals). For CDM channels, the frequency bandwidth of the carrier channel is much larger than the bandwidth of the original information signal. Because the channel bandwidth is very large, information from other channels operating in the same frequency band is relatively small. This allows multiple communications channels to operate in the same frequency bandwidth at the same time. There are various forms of CDM. The most popular forms of spread spectrum include frequency hopping and direct spread spectrum.

Frequency hopping is a multiplexing technology where transceivers may share a frequency band by transmitting for brief periods of time on an individual carrier channels and then hopping to other carrier channels to continue transmission. Each transceiver is assigned to a particular hopping pattern and collisions that occur are random. These errors only cause a loss of small amounts of data that may be fixed through error detection and correction methods.

Direct spread spectrum is relatively new commercialized (verses militarized) modulation technique that is used primarily in cellular and satellite systems. Direct sequence spread spectrum systems mix a relatively long digital code with a small amount of communication data (information signal) to produce a combined signal that is spread over a relatively wide frequency band. To receive the signal, the long code is used to extract the original signal.

Because the energy is spread over a wide bandwidth, multiple spread spectrum channels with different codes can co-exist with minimal interference. Figure 3.8 shows how a single direct sequence spread spectrum communication channel can have several channels. In this example, there are 3 different code patterns that are used for communication channels. When a receiver uses the reference code, a direct sequence spread spectrum system can build a mask as shown in Figure below for each conversation allowing only that information which falls within the mask to be transmitted or received.


Code Division Multiplexing


Digital Speech Interpolation (DSI)
In addition to multiplexing through channel division, statistical multiplexing can also be used by distributing transmission of a communications channel over idle portions of multiplexed channels. An example of statistical multiplexing is digital speech interpolation (DSI). DSI is a technique that dynamically allocates time slots for voice or data transmission to a user only when the have voice or data activity. This increases the system capacity as transmission for other users can occur when others are silent.

Digital speech interpolation (DSI) is a digital form of a process known as time assigned speech interpolation (TASI). The DSI technique that dynamically allocates channels (usually time slots) for voice or data transmission to a user only when the have voice or data activity. This increases the system capacity as transmission for other users can occur when others are silent.

A system that has DSI capability assigns information transmission based in speech activity. The DSI system senses activities of speech signals and availability of communication channels in a system and dynamically transmits information signals on available communications channels. Because speech conversation is composed of pauses and alternating directions of communications (usually one person speaks at a time), the use of TASI increases the efficiency of a communications system of approximately 2:1. For example, a 96 channel communications circuit that uses TASI can provide service approximately 192 calls.

Figure below shows the process of multiplexing using DSI. This diagram shows a communication circuit that has 96 independent communication channels (one communication link that has 96 time slots). The DSI system monitors the activity of each voice conversation (a voice channel) using a voice activity detector (VAD). The VAD is an electronic circuit that senses the activity (or absence) of voice signals. This is used to inhibit a transmission signal during periods of voice inactivity.


Digital Speech Interpolation (DSI)


When the VAD detects that speech is active, the DSI system assigns the information to specific time slots on the communications channel. The DSI transmitter system identifies the voice channel at the beginning of the transmission so the DSI receiver can assign it to an output voice channel. When the voice activity detector senses a pause in communication, the DSI transmitter sends an ending message on the channel allowing the channel to be placed back into a pool of available communication channels. The next time the speech activity detector senses the voice channel is again active, the DSI transmitter will select a channel from the pool of available communication channels and the process begins again. Each time, the DSI receiver will assign the information to the correct output voice channel.

Friday, March 7, 2008

Signaling : In-Band Signaling,

Signaling is the process of transferring control information such as connection addresses, call supervision codes, or other connection information between communication switching equipment and other communications equipment or systems. The basic functions of signaling include initiate a call or line connection (call setup), maintain a communication link, and to end a call or connection (call teardown). Signaling comes in two basic forms: in-band signaling and out-of-band signaling.

In-Band Signaling

In-band signaling sends control messages in the same communication channel that is used for voice or data communication. During the period of in-band signaling, the voice or data communication is temporarily inhibited (muted) to allow the transfer of control messages. The types of in-band address signaling include dial pulse (DP), dual-tone multi-frequency (DTMF), MF (Multi-Frequency), audio signaling, and line control.

Dial pulse (DP) signaling senses and counts the changes in current flow, such as from a rotary dial telephone, to allow the user to send address information (dialed digits) to the telephone system.

DTMF signaling is a means of transferring information from a user to the telephone network through the use of in-band audio tones. Each digit of information is assigned a simultaneous combination of one of a lower group of frequencies and one of a higher group of frequencies to represent each digit or character.

Multifrequency (MF) signaling is a type of in-band address signaling method that represents decimal digits and auxiliary signals by pairs of frequencies from the following group: 700, 900, 1100, 1300, 1500, and 1700 Hz. These audio frequencies are used to indicate telephone address digits, precedence, control signals, such as line-busy or trunk-busy signals, and other required signals.

On modern telephone systems, most in-band signaling only occurs between the end-user and his serving central office telephone switch.

These signals travel over the same audio line as the voice or data call. Examples of other these in-band signaling messages include:

- Dial tone (the circuit is working)

- Busy tone (the circuit is unavailable)

- Fast busy tone (the system is busy)

- DTMF or pulse digits (send dialed digit information)

- Special functions such as # and * (activate other services)

- Telephone systems also can sense line condition as a signaling method. When the central office senses a grounding of the line (ground start) or a reduction in voltage (off-hook loop-start), it produces a dialtone signal (audio signaling) to inform the user service is available. Wink start is another line activation signal that is used by the telephone switch to indicate to end-user telephone systems of a change in status. Wink signals are brief 140 msec interruptions of communication.

Out-of-Band Signaling
Out-of-band signaling is signaling that travels over a separate path from voice and data calls but carries control information about the calls such as call setup, call routing, caller-id, call tear-down, etc. For out-of-band signaling the telecommunication industry uses a standard called Common Channel Signaling and Control (CCS). The current version of CCS is known as Signaling System 7 (SS7). Through standardization all telephone companies implement SS7 and thus can interact smoothly with very few errors.

Using SS7, switches can more effectively route calls and even query centralized databases for additional control information. The advent of SS7 has brought with it many new features such as caller-id. It has also been instrumental allowing for the phenomenal growth the industry has seen.

Thursday, March 6, 2008

Virtual Circuits

A virtual circuit or virtual channel (VC) is a logical connection between two communication ports in one or more communication networks. There are two types of VC’s: permanent virtual circuits (PVC’s) and switched virtual circuits (SVC’s).

A SVC is an automatically and temporarily created virtual connection that is used for a communication session. A PVC is a virtual circuit is manually created for a continuous communication connection. To create a PVC, routing tables in switches are manually configured one time to provide a continuous connection of end points through a network. The ability to dynamically or manually create virtual connections through a network has created a new type of network referred to as value-added networks (VAN’s). Rather than purchase leased circuits between corporate locations some companies chose to contract with VAN’s to transport their data between their sites. With leased data circuits from each corporate site to the closest VAN point-of-presence (POP), usually in the same metropolitan area, a company could establish PVC’s between its sites through the VAN’s network. These functioned similar to leased circuits and often provided a monthly cost saving to the corporation, yet delivered practically the same service as leased data circuits.

Wednesday, March 5, 2008

Carrier System : Optical Carrier (OCx), ISDN Digital Subscriber Line (IDSL), Integrated Digital Loop Carrier (IDLC),

Optical carrier (OCx) transmission is a hierarchy of optical communication channels and lines that range from 51 Mbps to more than 39 Gbps. Lower level OC structures are combined to produce higher-speed communication lines. There are different structures of OC. The North American optical transmission standard is called synchronous optical network (SONET) and the European (world standard) is synchronous digital hierarchy (SDH). OCx has been used to represent the digital transmission standards where the “x” denotes the multiple of 51.84 Mbps service. Optical carrier standards continue up through OC768 but some definition at the higher levels is still lacking. The following is an abbreviated list of the optical carrier systems:

Optical Carrier 1 (OC1) – Operates at 51.84 Mbps;

Optical Carrier 3 (OC3) – Operates at 155.52 Mbps (3 X OC1)

Optical Carrier 9 (OC9) – Operates at 466.56 Mbps (9 X OC1);

Optical Carrier 12 (OC12) – Operates at 622.08 Mbps (12 X OC1).

Optical Carrier 192 (OC192) – Operates at 9.95 Gbps

Optical Carrier 256 (OC256) – Operates at 13.27 Gbps.

Optical Carrier 768 (OC768) – Operates at 39.81 Gbps

Synchronous Digital Hierarchy (SDH) is an international digital transmission format used in optical (fiber) networks standardized that is similar (but not identical) to the title Synchronous Optical Network (SONET) used in the United States. SDH uses standardized synchronous transmission according to CCITT standards G.707, G.708 and G.709. These standards define data transfer rates, defined optical interfaces and signal structure formats.

ISDN Digital Subscriber Line (IDSL)
ISDN Digital Subscriber Line (IDSL) is a hybrid of ISDN and DSL technologies. It uses the same data formatting as ISDN devices on the copper wire pair and delivers up to 144 kilobits per second bandwidth through two 64 kbps channels and one 16 kbps channel. The key difference for IDSL systems is that the IDSL system only uses the 64 kbps DS0 channels and the ISDN control channel (D channel) is ignored. The IDSL system effectively multiplies the number of channels on a single copper pair by 2x. The ability to avoid using ISDN signaling is very important as software upgrades for switching systems, to allow ISDN operation can cost more than $500,000 per switch.

Integrated Digital Loop Carrier (IDLC)
Integrated digital loop carrier (DLC) is a digital transmission technology that is used between the central office and groups of customers. The IDLC system is composed of two primary parts: an integrated digital terminal (IDT) and a remote digital terminal (RDT). The IDT concentrates up to 96 lines on to a single 24 channel T1 line. It does this by assigning central office channels to time slots on the IDLC line (between the IDT and RDT) as needed. The RDT reverses the process by assigning a time slot to an access line. The RDT also changes the format of the time slot to the access technology of choice (e.g., ISDN or analog).

The key advantages to the DLC carrier system is that some of the switching function is moved closer to the customer (in the RDT) and increased cost effective transmission through the increased sharing of local loop copper lines. Because the RDT in the DLC system acts as a repeater, this also extends the range of access lines from the central office to the end customer.

Unfortunately, DLC systems are not transparent to DSL systems. Although it is possible to install digital subscriber line network equipment (co-locate) along with RDT equipment, the RDT equipment housings and power supplies were not originally designed to hold additional equipment.

An RDT is divided into three major parts; digital transmission facility interface, common system interface, and line interface. The digital transmission interface terminates the high-speed line and coordinates the signaling. The common system interface performs the multiplexing/de-multiplexing, signaling insertion and extraction. The line interface contains digital to analog conversions (if the access line is analog) or digital formatting (if the line is digital).

Figure 4.23 shows an integrated digital loop carrier system. The DLC carrier system is composed of two basic parts: the Integrated Digital Terminal (IDT) and the Remote RDT. The IDT dynamically connects access lines (actually digital time slots) in the switching system to time slots on the communications line between the IDT and RDT. The RDT reverses the process and converts the high speed DLC channel into independent POTS channels (DS0s) that are connected via local loop lines to homes or businesses.


Figure 1: Digital Loop Carrier (DLC) System


The most common multiplexer is the SLC (subscriber line carrier)-96. This system, routinely referred to as a “Slick96”, can deliver 96 voice circuits (the equivalent 4 T-1’s) to the customer site. This methodology overcomes distance problems associated with providing voice services to remote customers as well as free additional twisted pair for future use.

Tuesday, March 4, 2008

Carrier System : Digital Subscriber Line (DSL)

Digital subscriber line is the transmission of digital information, usually on a copper wire pair. Although the transmitted information is in digital form, the transmission medium is usually an analog carrier signal (or the combination of many analog carrier signals) that is modulated by the digital information signal.

Digital subscriber line (DSL) was first used in the 1960s to describe the T-1 circuits that were extended to the customer premises. Later the same term was used to describe ISDN basic rate interface (BRI) (2B+D, 144 Kbps) and primary rates interface (PRI) (23B+D, 1.544 Mbps). There are several different digital subscriber line technologies. Each of these DSL technologies usually has a prefix to indicate the specific variant of DSL technology. Hence, the “x” in xDSL indicates that there are many forms of xDSL technology.

DSL transmission allows high-speed data transmission over existing twisted pair telephone wires. This has the potential providing high-speed data services without the burden of installing new transmission lines (e.g., for Internet access).

DSL service dramatically evolved in the mid 1990s due to the availability of new modulation technology and low cost electronic circuits that can do advanced signal processing (e.g., echo canceling and multiple channel demodulation). This has increased the data transmission capability of twisted pair copper wire to over 50 Mbps.

The data transmission capability of a DSL system varies based on the distance of the cable, type of cable used, and modulation technology. There are several different DSL technologies. Each of the DSL technologies mixes different types of transmission technologies to satisfy a specific business need. Some DSL systems allow simultaneous digital and analog transmission and are compatible with analog POTS systems.

Figure 1 shows a basic DSL system. This diagram shows that the key to DSL technologies is a more efficient use of the 1 MHz of bandwidth available on a single pair of copper telephone lines. A DSL system consists of compatible modems on each end of the local loop. For some systems, the DSL system allows for multiple types of transmission on a single copper pair. This includes analog or ISDN telephone (e.g., POTS) and digital communications (ADSL or VDSL). This diagram shows that there are basic trade offs for DSL systems. Generally, the longer the distance of the copper line, the lower the data rate. Distances of less than 1,000 feet can achieve data rates of over 50 Mbps.


Figure 1: Basic Digital Subscriber Line (DSL) System


The first digital subscriber lines (DSLs) were developed due to the need for cost effective quality communication over copper wire. The first digital transmission system was the T1 line. This system had a maximum distance of approximately 6,000 feet prior to needing repeaters.

The T1 digital transmission system used a very complex form of digital transmission. A new high-speed digital subscriber line technology was developed to replace T1 transmission technology. HDSL systems increased the distance that high-speed digital signals could be transmitted without the user of a repeater/amplifier. The HDSL system did require 2 (or 3) pairs of wires to allow simultaneous (send and receive) up to 2 Mbps of data transmission. To conserve the number of copper pairs for data transmission, symmetrical digital subscriber line (SDSL) technology was developed. Although SDSL systems offered lower data rates than HDSL, only 2 wire pairs were required. Since SDSL was developed, the HDSL system has evolved to a 2nd generation (HDSL2) that allows the use of 2 wire pair for duplex transmission with reduced emissions (lower egress). New efficient modulation technology used by ADSL systems dramatically increased the data transmission rates from the central office to the customer to over 6 Mbps (some ADSL systems to 8 Mbps). To take advantage of integrated services digital network (ISDN) equipment and efficiency, an offshoot of ISDN technology that was adapted for the local loop developed called ISDN digital subscriber line (IDSL). Asymmetric digital subscriber line (ADSL) systems evolved to rate adaptive digital subscriber line (RADSL) allow the data rate to be automatically or manually changed by the service provider. To simplify the installation of consumer based DSL equipment, and low data transmission offshoot of ADSL developed that is called ADSL-Lite. Using similar technology as the ADSL system, very high-speed digital subscriber line (VDSL) was created to provide up to 52 Mbps data transfer rates over very short distances.

Figure 2 shows the evolution of DSL systems. This diagram shows that high-speed digital subscriber line technology has been readily available since the 1970s. In the late 1990’s, the addition of advanced signal processing technology allowed DSL technology to rapidly increase transmission speed to over 50 Mbps in short distances.


Figure 2: Evolution of DSL

Digital Subscriber Line (DSL)

Monday, March 3, 2008

Carrier System : Integrated Services Digital Network (ISDN)

Integrated services digital network (ISDN) is a structured all digital telephone network system that was designed to replace (upgrade) existing analog telephone networks. The ISDN network supports for advanced telecommunications services and defined universal standard interfaces that are used in wireless and wired communications systems. There are two key user interfaces defined for ISDN networks: basic rate interface (BRI) and primary rate interface (PRI).

The basic rate interface (BRI) is the smallest transmission system (or interface) available through ISDN. BRI provides for two 64 kbps bearer channels (B channels) and a 16 kbps signaling (data) channel (D channel). This configuration is also is also referred to as 2B+D.

The primary rate interface (PRI) is a standard high-speed data communications interface that is used in the ISDN system. This interface provides a standard data rates for T1 1.544 Mbps and E1 2.048 Mbps. The interface can be divided into combinations of 384 kbps (H) channels, 64 kbps (B) channels and includes at least one 64 kbps (D) control channel.

Integrated Services Digital Network (ISDN)

Saturday, March 1, 2008

Carrier System : Digital Signal Level (DSx)

Digital signal level (DSx) transmission is a hierarchy of digital communication channels and lines that range from 64 kbps to 565 Mbps. Lower level DS structures are combined to produce higher-speed communication lines. There are different structures of DS levels used throughout the world with significant variations between North American and European systems. DSx has been used to represent the digital transmission standards where the “x” denotes which service is under discussion.

Trunk carrier (T Carrier) is often used to describe the DSx level. Trunk carrier uses (Tx) to represent the digital transmission standards where the “x” denotes the multiplexed level of trunk service. Tx is the actual transmission structure where DSx is the digital signal levels. Outside the United States, E Carrier (Ex) is used to represent the transmission carrier.

The following represent the North American DS structure:

Digital Signal 0 (DS0) - is the smallest digital channel operating at 64 kbps. It represents a single digitized analog voice channel;

Digital Signal 1 (DS1)
– normally referred as a T-1, it is composed of twenty four voice channels packed into a 193 bit frame and transmitted at 1.544 Mbps. The unframed version, or payload, is 192 bits at a rate of 1.536 Mbps;

Digital Signal 2 (DS2) – normally referred to as a T-2, it is composed of four T-1 frames packed into a higher level frame transmitted at 6.312 Mbps;

Digital Signal 3 (DS3) – normally referred to as a T-3, it is composed of twenty-eight T-1 frames packed into a level frame transmitted at 44.736 Mbps.

Fractional T1 (FracT) - A data transmission rate that is a portion of the total capacity of a T-1 communications line (1.544 Mbps) but greater than a DS0 (64Kbps).

European standards are different from the North American noted above. The European hierarchy is as follows:

Digital Signal 0 (DS0) - is the smallest digital channel operating at 64 kbps. It represents a single digitized analog voice channel;

Digital Signal 1 (DS1) – normally referred as an E-1, it is composed of thirty voice channels and two controls channels. It uses a 256-bit frame and operates at 2.048 Mbps.

Digital Signal 2 (DS2) – normally referred to as a E-2, it is composed of four E-1 frames packed into a higher level frame transmitted at 8.448 Mbps;

Digital Signal 3 (DS3) – normally referred to as an E-3, it is composed of sixteen E-1 frames packed into a level frame transmitted at 34.368 Mbps.

Digital Signal 4 (DS4) – normally referred to as a E-4, it is composed of sixty-four E-1 frames packed into a higher level frame transmitted at 139.268 Mbps;

Digital Signal 5 (DS5) – normally referred to as an E-5, it is composed of two hundred fifty-six E-1 frames packed into a level frame transmitted at 565.148 Mbps.

Digital Signal Level (DSx)