Overview,
Transmission systems interconnect communication devices (end nodes) by guiding signal energy in a particular direction or directions through a transmission medium such as copper, air, or glass. A transmission system will have at least one transmitting device, a transmission medium, and a receiving device. The transmitting communication device is capable of converting an information signal into a form of electrical, electromagnetic wave (radio), or optical signal that allow the information to be transferred through the transmission medium. The receiving communication device converts the transmitted signal into another form that can be used by the device or other devices that are connected to it. Transmission systems can be unidirectional (one direction) or they can be bi-directional (two directions).
The basic types of transmission mediums include copper wire, coaxial cable, free space/air, fiber optic cable, and mechanical transmission line. Copper and coaxial wire is primarily used for low to moderate frequency transmission over a few miles. Free space/air systems can transmit hundreds of miles but have limited bandwidth and are susceptible to noise interference. Fiber optic cable is capable of carrying high-speed data signals (as light pulses) over thousands of miles. Mechanical (acoustic wave) transmission lines transmit over very short distances (only a few millimeters) and are used for signal filtering components.
Different types of transmission lines have varying performance characteristics and may be susceptible to interference during signal transmission. These characteristics include the available frequency bandwidth (frequency response), how much signal leakage may occur (cross talk), and the susceptibility of absorbing other signals (signal ingress). The construction of the transmission line itself may cause distortions in the transmitted signal. This includes unterminated line splices (bridge tap reflections), poor line splices, and line resistance (signal attenuation). Other characteristics such as varying delays to different frequency ranges may cause group dispersion (smearing) of the desired signal.
To allow devices to communicate with each other over a transmission line, carrier systems specify the signal types and levels along with specific protocol controls (communication rules). These carrier systems are often specific to the transmission medium such as copper or fiber. Some of the more popular carrier systems include plain old telephone service (POTS), digital signaling carrier (DSx), digital subscriber line (DSL), and optical carrier (OCx).
To coordinate the transmission line, signaling messages are sent between communication devices. Some of these control messages are sent along with the data on the transmission line (called in-band signaling) and others are sent through another path or network (called out-of-band signaling).
In some cases, a transmission path may only be a portion of a path (a logical path) through a transmission line. The length of a transmission may be extended through the use of amplifiers or repeaters.
Friday, February 22, 2008
Thursday, February 21, 2008
Simple Telecom : Security
Security management of a network involves identity validation (authentication), service authorization, and information privacy protection. Authentication processes identifies the device or person that is requesting the use of the telecommunications device or network services. Authorization is the process of determining what services devices are customers are permitted to use. Privacy or encryption services are used to help ensure that the information transmitted or received is not available to unauthorized recipients.
Authentication
Authentication is a process during where information is exchanged between a communications device (typically a user device such as a mobile phone) and a communications network that allows the carrier or network operator to confirm the true identity of the user (or device). Validation of the authenticity of the user or device allows a service provider to deny service to users that cannot be identified. Thus, authentication inhibits fraudulent use of a communication device that does not contain the proper identification information.
Authorization
Authorization is the enabling of services to a device or customer that requests services. Authorization is often part of the billing and customer care (BCC) system and is maintained in a customer database service profile. Services are initially enabled in a network as a result of provisioning. Provisioning is a process within a company that allows for establishment of new accounts, activation, termination of features, and coordinating and dispatching the resources necessary to fill those service orders. Provisioning is usually part of customer care systems.
Networks sometimes use mediation devices to help manage provisioning and authorizing services to customers. A mediation device is device in a telecommunications network that receives, processes, reformats, and sends information to other formats between network elements. Mediation devices are can take non-standard proprietary information (such as proprietary digital call detail records) from switches and other network equipment and reformat them into messages billing systems can understand.
Information Privacy
Information privacy is a process of protecting transmitted or received information from being understood by unauthorized recipients. Information privacy typically involves encrypting of the voice signal with a shared secret key so only authorized users with the correct key and decryption program can listen to the communication information.
Encryption is a process of a protecting voice or data information from being obtained by unauthorized users. Encryption involves the use of a data processing algorithm (formula program) that uses one or more secret keys that both the sender and receiver of the information use to encrypt and decrypt the information. Without the encryption algorithm and key(s), unauthorized listeners cannot decode the message.
Authentication
Authentication is a process during where information is exchanged between a communications device (typically a user device such as a mobile phone) and a communications network that allows the carrier or network operator to confirm the true identity of the user (or device). Validation of the authenticity of the user or device allows a service provider to deny service to users that cannot be identified. Thus, authentication inhibits fraudulent use of a communication device that does not contain the proper identification information.
Authorization
Authorization is the enabling of services to a device or customer that requests services. Authorization is often part of the billing and customer care (BCC) system and is maintained in a customer database service profile. Services are initially enabled in a network as a result of provisioning. Provisioning is a process within a company that allows for establishment of new accounts, activation, termination of features, and coordinating and dispatching the resources necessary to fill those service orders. Provisioning is usually part of customer care systems.
Networks sometimes use mediation devices to help manage provisioning and authorizing services to customers. A mediation device is device in a telecommunications network that receives, processes, reformats, and sends information to other formats between network elements. Mediation devices are can take non-standard proprietary information (such as proprietary digital call detail records) from switches and other network equipment and reformat them into messages billing systems can understand.
Information Privacy
Information privacy is a process of protecting transmitted or received information from being understood by unauthorized recipients. Information privacy typically involves encrypting of the voice signal with a shared secret key so only authorized users with the correct key and decryption program can listen to the communication information.
Encryption is a process of a protecting voice or data information from being obtained by unauthorized users. Encryption involves the use of a data processing algorithm (formula program) that uses one or more secret keys that both the sender and receiver of the information use to encrypt and decrypt the information. Without the encryption algorithm and key(s), unauthorized listeners cannot decode the message.
Wednesday, February 20, 2008
Telecom : Network Control
Network control is the transmission of signals or messages that perform call control, equipment configuration, or information management functions. Network control can be centralized or distributed. The control of public telecommunications networks is a centralized system as call processing is coordinated through a controlled common channel signaling (CCS) network. The Internet uses distributed control as the switching information dynamically changes in packet switching centers (routers) throughout the Internet network.
Common Channel Signaling (CCS)
Common channel signaling system #7 (“SS7”) is the primary system used for interconnection of telephone systems. SS7 sends packets of control information between switching systems. Figure below shows the basic structure of the SS7 control signaling system. The SS7 network is composed of its own data packet switches, and these switching facilities are called signal transfer points (STPs). In some cases, when advanced intelligent network services are provided, STPs may communicate with signal control points (SCPs) to process advanced telephone services. STPs are the telephone network switching point that route control messages to other switching points. SCPs are databases that allow messages to be processed as they pass through the network (such as calling card information or call forwarding information).
SS7 Common Channel Signaling
Because the public telephone network uses common channel signaling, intelligence in the network can be distributed to databases and information processing points throughout the network. A set of service development tools has been developed to allow companies to offer advanced intelligent network (AIN) services.
Common Channel Signaling (CCS)
Common channel signaling system #7 (“SS7”) is the primary system used for interconnection of telephone systems. SS7 sends packets of control information between switching systems. Figure below shows the basic structure of the SS7 control signaling system. The SS7 network is composed of its own data packet switches, and these switching facilities are called signal transfer points (STPs). In some cases, when advanced intelligent network services are provided, STPs may communicate with signal control points (SCPs) to process advanced telephone services. STPs are the telephone network switching point that route control messages to other switching points. SCPs are databases that allow messages to be processed as they pass through the network (such as calling card information or call forwarding information).
Because the public telephone network uses common channel signaling, intelligence in the network can be distributed to databases and information processing points throughout the network. A set of service development tools has been developed to allow companies to offer advanced intelligent network (AIN) services.
Tuesday, February 19, 2008
Telecom : Protocols
Protocols are the precise set of rules and a syntax that govern the accurate transfer of information within a communications network. Protocols are used within a communication system to establish, carry out, and terminate communication circuits. Protocols are also used to coordinate billing and customer care systems, manage network devices, and any other process that requires coordinated communication and control.
There are thousands of different protocols used in communications systems. Usually, protocols are grouped into families of protocols so they can serve specific types of networks and services. When interconnecting different networks, protocols need to be converted.
Protocol conversion involves the translation of the protocols of one system to those of another to enable different types of equipment, such as data terminals and computers, to communicate. This is done by an inter-working function (IWF). An IWF system (such as a data bridge) adapts the communications between two different types of networks. Protocol conversion may be used to interconnect circuit switched or packet switched networks.
There are thousands of different protocols used in communications systems. Usually, protocols are grouped into families of protocols so they can serve specific types of networks and services. When interconnecting different networks, protocols need to be converted.
Protocol conversion involves the translation of the protocols of one system to those of another to enable different types of equipment, such as data terminals and computers, to communicate. This is done by an inter-working function (IWF). An IWF system (such as a data bridge) adapts the communications between two different types of networks. Protocol conversion may be used to interconnect circuit switched or packet switched networks.
Monday, February 18, 2008
Switching Systems
Switching systems connect two (or more) points together. These connections can be physically connected (mechanical switch) or connected logically (through software).
The first telephone systems performed the mechanical switching of calls by human operators. The operators interconnected telephone lines by manually connecting cables at switchboards. These switchboards contained many wires that had plugs and the switchboard had many sockets for the plugs. To interconnect telephone calls at long distances, one operator would have to call other operators to setup the call. Setting up calls could be a complex process and this process got more complex as many more telephones were installed.
Switching systems have evolved many times over the past 100 years. The types of switching systems that are still in common use today include crossbar, time slot interchange (TSI), and packet switching.
Crossbar
Crossbar switches used mechanical arms to physically connect to wires (or busses) together. These mechanical arms (“Crossbars”) connect horizontal and vertical bars together to connect input and output lines together. Magnets are used to open and close the crossbar switch contacts.
Figure below shows a crossbar switching system. In this example, there is a matrix of lines (busses) where each input line can be connected to any output line. When a connection needs to be made, a mechanical switch connects one of the busses with the other busses. The disadvantage of this system is that the number of mechanical switches for connecting each input port to an output port exponentially increases with the number of ports that require connection. For example, a switch with 10 inputs and 10 output lines requires 100 switches. A switch that has 20 inputs and 20 outputs requires 400 switches.
Crossbar Switching
Time Slot Interchange (TSI)
Time slot interchange (TSI) switching is a process of connecting incoming and outgoing digital lines together through the use of temporary memory locations. In the late 1960’s, mechanical crossbar switching systems began to change to TSI digital switching systems. A computer controls the assignment of these temporary locations so that a portion of an incoming line can be stored in temporary memory and retrieved for insertion to an outgoing line.
Figure below shows a TSI switching system. This diagram shows a simplified matrix switching system. Each input line (port) is connected to a multiplexer. The multiplexer places data from each port in time sequence (time slot) on a communications line (e.g., a T1 or E1 line). This time multiplexed signal is supplied to a matrix switching assembly. The matrix switching assembly core has two memory parts: a section that holds the pulse coded modulation (PCM) data and Control Memory - CRAM that holds switching addresses data.
Time Slot Interchange (TSI) Switching
The time slots (voice channels) from the incoming multiplexed sent through switch S1 to be sequentially stored in the PCM data memory. The data is later retrieved by switch S2 and placed on a specific time slot on an outgoing line. The outgoing multiplexed line is supplied to a de-multiplexer so each time slot is routed to an output port.
Packet Switches
Packet transmission is a mode of data transmission that divides messages or data into small increments (packets) that can be routed through a network. When the packets arrive at their destination, they are reassembled in the proper order to recreate the original message or data.
Packet switching can be connection based or connectionless. For connection based switching, a path through the network is established during call initiation and packets are continuously routed through the same path. For connectionless switching, each packet is given a destination address and the switching points in the network (switching nodes) assist in routing the packet to its destination.
Figure below shows two types of packet switching in a communications system. Diagram (a) shows that connection based packet switching sets up a communication circuit prior to transmitting packets that contain data. Diagram (b) shows connectionless packet switching. Connectionless packet switching requires intelligent switching nodes (routers) that can decode the destination address and select the forwarding route based on the results of the lookup in the routing table. This diagram shows that packets of data arrive at the switch. The routing switch extracts the destination address and possibly the type of message.
Packet Switches
The first telephone systems performed the mechanical switching of calls by human operators. The operators interconnected telephone lines by manually connecting cables at switchboards. These switchboards contained many wires that had plugs and the switchboard had many sockets for the plugs. To interconnect telephone calls at long distances, one operator would have to call other operators to setup the call. Setting up calls could be a complex process and this process got more complex as many more telephones were installed.
Switching systems have evolved many times over the past 100 years. The types of switching systems that are still in common use today include crossbar, time slot interchange (TSI), and packet switching.
Crossbar
Crossbar switches used mechanical arms to physically connect to wires (or busses) together. These mechanical arms (“Crossbars”) connect horizontal and vertical bars together to connect input and output lines together. Magnets are used to open and close the crossbar switch contacts.
Figure below shows a crossbar switching system. In this example, there is a matrix of lines (busses) where each input line can be connected to any output line. When a connection needs to be made, a mechanical switch connects one of the busses with the other busses. The disadvantage of this system is that the number of mechanical switches for connecting each input port to an output port exponentially increases with the number of ports that require connection. For example, a switch with 10 inputs and 10 output lines requires 100 switches. A switch that has 20 inputs and 20 outputs requires 400 switches.
Time Slot Interchange (TSI)
Time slot interchange (TSI) switching is a process of connecting incoming and outgoing digital lines together through the use of temporary memory locations. In the late 1960’s, mechanical crossbar switching systems began to change to TSI digital switching systems. A computer controls the assignment of these temporary locations so that a portion of an incoming line can be stored in temporary memory and retrieved for insertion to an outgoing line.
Figure below shows a TSI switching system. This diagram shows a simplified matrix switching system. Each input line (port) is connected to a multiplexer. The multiplexer places data from each port in time sequence (time slot) on a communications line (e.g., a T1 or E1 line). This time multiplexed signal is supplied to a matrix switching assembly. The matrix switching assembly core has two memory parts: a section that holds the pulse coded modulation (PCM) data and Control Memory - CRAM that holds switching addresses data.
The time slots (voice channels) from the incoming multiplexed sent through switch S1 to be sequentially stored in the PCM data memory. The data is later retrieved by switch S2 and placed on a specific time slot on an outgoing line. The outgoing multiplexed line is supplied to a de-multiplexer so each time slot is routed to an output port.
Packet Switches
Packet transmission is a mode of data transmission that divides messages or data into small increments (packets) that can be routed through a network. When the packets arrive at their destination, they are reassembled in the proper order to recreate the original message or data.
Packet switching can be connection based or connectionless. For connection based switching, a path through the network is established during call initiation and packets are continuously routed through the same path. For connectionless switching, each packet is given a destination address and the switching points in the network (switching nodes) assist in routing the packet to its destination.
Figure below shows two types of packet switching in a communications system. Diagram (a) shows that connection based packet switching sets up a communication circuit prior to transmitting packets that contain data. Diagram (b) shows connectionless packet switching. Connectionless packet switching requires intelligent switching nodes (routers) that can decode the destination address and select the forwarding route based on the results of the lookup in the routing table. This diagram shows that packets of data arrive at the switch. The routing switch extracts the destination address and possibly the type of message.
Saturday, February 16, 2008
Control Message Signaling [Simple Telecom]
Control message signaling (commonly called “signaling”) is the process of transferring control information such as address, call supervision, or other connection information between communication equipment and other equipment or systems. There are two methods used for signaling: in-band and out-of band signaling.
In-Band Signaling
In-band signaling occurs when control messages share the same communication channel as the information signals (e.g., within the audio signal bandwidth). In-band signaling requires the users voice or data information to be momentarily interrupted or altered while signaling messages are being transferred. In-band signaling is sometimes called blank and burst signaling.
Figure below shows the process of in-band signaling. This diagram shows that a signaling message has been created to control the communications line (e.g., to transfer a call). To allow the transmission of the control message, the information is temporarily inhibited (or discarded) and the control message is sent on the same channel.
In-Band Signaling
Out-of-Band Signaling
Out-of-band signaling is a process of sending control signals outside of the communication channel that is in use (e.g., outside the audio signal frequency range). Out-of-band signaling allows uninterrupted communication while the users voice or data information is being transferred.
Figure below shows how out-of-band signaling occurs. This diagram shows that a control message can either be sent on the same channel but in different time slots than the information (e.g., voice) signal or over a separate control signaling network (called common channel signaling).
Out-of-Band Signaling
In-Band Signaling
In-band signaling occurs when control messages share the same communication channel as the information signals (e.g., within the audio signal bandwidth). In-band signaling requires the users voice or data information to be momentarily interrupted or altered while signaling messages are being transferred. In-band signaling is sometimes called blank and burst signaling.
Figure below shows the process of in-band signaling. This diagram shows that a signaling message has been created to control the communications line (e.g., to transfer a call). To allow the transmission of the control message, the information is temporarily inhibited (or discarded) and the control message is sent on the same channel.
Out-of-Band Signaling
Out-of-band signaling is a process of sending control signals outside of the communication channel that is in use (e.g., outside the audio signal frequency range). Out-of-band signaling allows uninterrupted communication while the users voice or data information is being transferred.
Figure below shows how out-of-band signaling occurs. This diagram shows that a control message can either be sent on the same channel but in different time slots than the information (e.g., voice) signal or over a separate control signaling network (called common channel signaling).
Friday, February 15, 2008
Digital Signal Processing
Digital signals processing refers to a category of electronic devices that represent and process information that are in discrete signal level (digital) formats. Digital signal processing refers to the manipulation of digital signals to change their content and to add error detection and correction capability.
Digital signals typically vary in two levels; on (logic 1) and off (logic 0). A bit is the smallest part of a digital signal, typically called a data bit. A bit typically can assume two levels: either a zero (0) or a one (1). A byte is an agreed-upon group of bits, typically eight. A byte typically represents one alphabetic or special character, two decimal digits, or eight binary bits of information.
When analog signals are converted to digital format, the digital signals represent the original analog waveform. Just like analog signals that may be processed by filters, shaping circuits, combiners and amplifiers, digital signals can be processed to produce similar functions. However, because the signal is in digital form, these functions are performed by software programs that manipulate the data.
Unlike analog signals, digital signals can be recreated to their original form. This process is called signal regeneration. To increase the efficiency of a transmission signal (allow more users per channel), digital signals can be analyzed for redundancy and the digital signal data can be compressed. Digital signals can also be processed in a way that helps overcome the effects signal distortion that can result in the incorrect determination of a digital signal (whether a zero or one had been sent). This is called error detection and error correction processing. When digital signals represent the original analog signal, advanced echo canceling software programs can be used to reduce the effects of echoes that are caused by feedback in the audio and transmission system. Some systems use dedicated digital signal processors (DSPs) to manipulate the incoming digital information via a program (stored instructions) that produce a new digital output. This allows software programs to perform many functions (such as signal filtering) that previously required complex dedicated electronic circuits.
Digitization of an Analog Signal
Analog signals must be converted to digital form for use in a digital wireless system. To convert analog signals to digital form, the analog signal is digitized by using an analog-to-digital (pronounced A to D) converter. The A/D converter periodically senses (samples) the level of the analog signal and creates a binary number or series of digital pulses that represent the level of the signal.
The common conversion process is Pulse Code Modulation (PCM). For most PCM systems, the typical analog sampling rate occurs at 8000 times a second. Each sample produces 8 bits digital that results in a digital data rate (bit stream) of 64 thousand bits per second (kbps).
Figure below shows how an analog signal is converted to a digital signal. This diagram shows that an acoustic (sound) signal is converted to an audio electrical signal (continuously varying signal) by a microphone. This signal is sent through an audio band-pass filter that only allows frequency ranges within the desired audio band (removes unwanted noise and other non-audio frequency components). The audio signal is then sampled every 125 microseconds (8,000 times per second) and converted into 8 digital bits. The digital bits represent the amplitude of the input analog signal.
Signal Digitization
Digital bytes of information are converted to specific voltage levels based on the value (weighting) of the binary bit position. In the binary system, the value of the next sequential bit is 2 times larger. For PCM systems that are used for telephone audio signals, the weighting of bits within a byte of information (8 bits) is different than the binary system. The companding process increases the dynamic range of a digital signal that represents an analog signal; smaller bits are given larger values that than their binary equivalent. This skewing of weighing value give better dynamic range. This companding process increases the dynamic range of a binary signal by assigning different weighted values to each bit of information than is defined by the binary system.
Two common encoding laws are Mu-Law and A-Law encoding. Mu-Law encoding is primarily used in the Americas and A-Law encoding is used in the rest of the world. When different types of encoding systems are used, a converter is used to translate the different coding levels.
Digital Signal Regeneration
To overcome the effects of noise on transmitted signals, digital transmission systems use digital signal regeneration to restore the quality of the signal as it moves through a network. Digital signal regeneration is the process of reception and restoration of a digital pulse or lightwave signal to its original form after its amplitude, waveform, or timing have been degraded by normal factors during transmission. The resultant signal is virtually free of noise or distortion.
Figure below shows the process of digital signal regeneration. This example shows an original digital signal (a) and added noise (b) to produce a combined digital signal with noise (c). The regeneration process detects maximum and minimum expected values (threshold points) and recreates the original digital signal (d).
Digital Signal Regeneration
Data Compression
To increase the amount of information that a transmission system can transfer, digital systems may use data compression. Data compression is a processing technique for encoding information so that fewer data bits of information are required to represent a given amount of data. Compression allows the transmission of more data over a given amount of time and circuit capacity. It also reduces the amount of memory required for data storage.
Digital compression analyzes a digital signal for either redundant information (repeated 1’s or 0’s) or may analyze the information content of the digital signal into component parts (such as speech patterns or video frames). All of this processing allows the data transmission rate to be reduced by sending only the characteristics of the signal rather than the complete digital signal. Some data compression systems can only reduce data rates by a factor of 2:1 (e.g., ADPCM audio compression) while advanced digital audio compression can only reduce data rates by a factor of approximately 200:1 (e.g., MPEG video compression). When used in combination of data compression and decompression, the device is called a COder/DECoder (CoDec).
When a digital signal is compressed for voice communications, it is called a voice coder (Vo-coder) or speech coder. The Vo-coder is a digital signal processing device that analyzes speech signals so that it can produce a lower data rate compressed digital signal. The difference between standard data compression and voice data compression is the analysis of the information source (speech) and elimination of compression process for non-voice signals. Speech coding usually involves the use of data tables (called code books) that represent information parts that can be associated with human sound. Because non-human sounds can be eliminated from the code book, this allows the number of bits can be used to create a compressed digital voice signal to be reduced.
Figure below shows the digital voice compression process. In this example, a digital signal is continuously applied to a digital signal analysis device. The analysis portion of the speech coder extracts the amplitude, pitch, and other key parameters of the signal and then looks up related values in the code book for the portion of sound it has analyzed. Only key parameters and code book values are transmitted. This results in data compression ratios of 4:1 to over 16:1.
Digital Voice Compression
Error Detection and Error Correction
To help reduce the effects of errors on data transmission, error detection, and error protection systems are used in most communication systems. Error detection systems use a process of adding some data bits to the transmitted data signal that are used to help determine if bits were received in error due to distorted transmission. Error correction is made possible by sending bits that have a relationship to the data that is contained in the desired data block or message. These related bits permit a receiver of information to use these extra information bits to detect and/or correct for errors that may have occurred during data transmission.
A common measurement of the performance of a communication system is the amount of bits received in error, called the bit error rate (BER). The BER is the ratio of bits received in error compared to the total number of bits received.
Error detection processing involves the creation of additional bits that are sent with the original data. The additional check bits are created by using a formula calculation on the digital signal prior to sending the data. After the digital signal is received, the formula can be used again to create check bits from the received digital signal. If the check bits match, the original digital signal was received correctly. If the check bits do not match, some (or all) of the digital signal was received in error. This process is called error detection.
Some digital systems use sophisticated mathematical formulas to create the check bits so that the check bits can be used to make corrections (or predictions of the correct bits) to the received digital signal. This process is called error correction.
Figure below shows the basic error detection and correction process. This diagram shows that a sequence of digital bits is supplied to a computing device that produces a check bit sequence. The check bit sequence is sent in addition to the original digital bits. When the check bits are received, the same formula is used to check to see if any of the bits received were in error.
Error Detection and Correction
Echo Cancellation
Echo cancellation is a process of extracting a delayed version of an original transmitted signal (audio echo) from the received signal. Echoes may be created through acoustic feedback where some of the audio signal transferring from a speaker into a microphone.
Echoed signals cause distortion and may be removed by performing via advanced signal analysis and filtering. Figure 3.18 shows an example of the echo cancellation process. This diagram shows how the combining of two signals, the original plus a delayed version of the original produces a complex signal. The echo canceling system analyzes the complex signal and uses the signal analysis to create variations of the likely echo signal. This prediction of echoed signal is subtracted from the complex signal to reproduce the original signal without the echo.
Echo Cancellation
Echoed signals can also occur in signals other than audio signals. When echoes occur on radio channels (the broadband signal), it is usually the result of the same signal that travels on different paths to reach its destination. This is called multipath propagation. Echo canceling can be used to reduce the effects of radio multipath propagation.
Digital Signal Processor (DSP)
A digital signal processor (DSP) is an electronics device or assembly (typically an integrated circuit) that is designed to process signals through the use of embedded microprocessor instructions. The use of DSPs in communication circuits allows manufacturers to quickly and reliably develop advanced communications systems through the use of software programs. The software programs (often called modules) perform advanced signal processing functions that previously complex dedicated electronics circuits. Although manufacturers may develop their own software modules, DSP software modules are often developed by other companies that specialize in specific types of communication technologies. For example, a manufacturer may purchase a software module for echo canceling from one DSP software module developer and a modulator software module from a different DSP software module developer. Because DSPs use these software modules, if new technologies such as speech compression, channel coding, or modulation techniques are developed, the manufacturer only has to change the software programs in the DSP to utilize the new technology.
Figure below shows typical digital signal processor that is used in a digital communication system. This diagram shows that a DSP contains a signal input and output lines, a microprocessor assembly, interrupt lines from assemblies that may require processing, and software program instructions. This diagram shows that this DSP has 3 software programs, digital signal compression, channel coding, and modulation coding. The digital signal compression software analyzes the digital audio signal and compresses the information to a lower data transmission rate. The channel coding adds control signals and error protection bits. The modulation coding formats (shapes) the output signal so it can be directly applied to an RF modulator assembly. This diagram also shows that an optional interface is included to allow updating of the software programs that are stored in the DSP.
Digital Signal Processor (DSP)
Digital signals typically vary in two levels; on (logic 1) and off (logic 0). A bit is the smallest part of a digital signal, typically called a data bit. A bit typically can assume two levels: either a zero (0) or a one (1). A byte is an agreed-upon group of bits, typically eight. A byte typically represents one alphabetic or special character, two decimal digits, or eight binary bits of information.
When analog signals are converted to digital format, the digital signals represent the original analog waveform. Just like analog signals that may be processed by filters, shaping circuits, combiners and amplifiers, digital signals can be processed to produce similar functions. However, because the signal is in digital form, these functions are performed by software programs that manipulate the data.
Unlike analog signals, digital signals can be recreated to their original form. This process is called signal regeneration. To increase the efficiency of a transmission signal (allow more users per channel), digital signals can be analyzed for redundancy and the digital signal data can be compressed. Digital signals can also be processed in a way that helps overcome the effects signal distortion that can result in the incorrect determination of a digital signal (whether a zero or one had been sent). This is called error detection and error correction processing. When digital signals represent the original analog signal, advanced echo canceling software programs can be used to reduce the effects of echoes that are caused by feedback in the audio and transmission system. Some systems use dedicated digital signal processors (DSPs) to manipulate the incoming digital information via a program (stored instructions) that produce a new digital output. This allows software programs to perform many functions (such as signal filtering) that previously required complex dedicated electronic circuits.
Digitization of an Analog Signal
Analog signals must be converted to digital form for use in a digital wireless system. To convert analog signals to digital form, the analog signal is digitized by using an analog-to-digital (pronounced A to D) converter. The A/D converter periodically senses (samples) the level of the analog signal and creates a binary number or series of digital pulses that represent the level of the signal.
The common conversion process is Pulse Code Modulation (PCM). For most PCM systems, the typical analog sampling rate occurs at 8000 times a second. Each sample produces 8 bits digital that results in a digital data rate (bit stream) of 64 thousand bits per second (kbps).
Figure below shows how an analog signal is converted to a digital signal. This diagram shows that an acoustic (sound) signal is converted to an audio electrical signal (continuously varying signal) by a microphone. This signal is sent through an audio band-pass filter that only allows frequency ranges within the desired audio band (removes unwanted noise and other non-audio frequency components). The audio signal is then sampled every 125 microseconds (8,000 times per second) and converted into 8 digital bits. The digital bits represent the amplitude of the input analog signal.
Digital bytes of information are converted to specific voltage levels based on the value (weighting) of the binary bit position. In the binary system, the value of the next sequential bit is 2 times larger. For PCM systems that are used for telephone audio signals, the weighting of bits within a byte of information (8 bits) is different than the binary system. The companding process increases the dynamic range of a digital signal that represents an analog signal; smaller bits are given larger values that than their binary equivalent. This skewing of weighing value give better dynamic range. This companding process increases the dynamic range of a binary signal by assigning different weighted values to each bit of information than is defined by the binary system.
Two common encoding laws are Mu-Law and A-Law encoding. Mu-Law encoding is primarily used in the Americas and A-Law encoding is used in the rest of the world. When different types of encoding systems are used, a converter is used to translate the different coding levels.
Digital Signal Regeneration
To overcome the effects of noise on transmitted signals, digital transmission systems use digital signal regeneration to restore the quality of the signal as it moves through a network. Digital signal regeneration is the process of reception and restoration of a digital pulse or lightwave signal to its original form after its amplitude, waveform, or timing have been degraded by normal factors during transmission. The resultant signal is virtually free of noise or distortion.
Figure below shows the process of digital signal regeneration. This example shows an original digital signal (a) and added noise (b) to produce a combined digital signal with noise (c). The regeneration process detects maximum and minimum expected values (threshold points) and recreates the original digital signal (d).
Data Compression
To increase the amount of information that a transmission system can transfer, digital systems may use data compression. Data compression is a processing technique for encoding information so that fewer data bits of information are required to represent a given amount of data. Compression allows the transmission of more data over a given amount of time and circuit capacity. It also reduces the amount of memory required for data storage.
Digital compression analyzes a digital signal for either redundant information (repeated 1’s or 0’s) or may analyze the information content of the digital signal into component parts (such as speech patterns or video frames). All of this processing allows the data transmission rate to be reduced by sending only the characteristics of the signal rather than the complete digital signal. Some data compression systems can only reduce data rates by a factor of 2:1 (e.g., ADPCM audio compression) while advanced digital audio compression can only reduce data rates by a factor of approximately 200:1 (e.g., MPEG video compression). When used in combination of data compression and decompression, the device is called a COder/DECoder (CoDec).
When a digital signal is compressed for voice communications, it is called a voice coder (Vo-coder) or speech coder. The Vo-coder is a digital signal processing device that analyzes speech signals so that it can produce a lower data rate compressed digital signal. The difference between standard data compression and voice data compression is the analysis of the information source (speech) and elimination of compression process for non-voice signals. Speech coding usually involves the use of data tables (called code books) that represent information parts that can be associated with human sound. Because non-human sounds can be eliminated from the code book, this allows the number of bits can be used to create a compressed digital voice signal to be reduced.
Figure below shows the digital voice compression process. In this example, a digital signal is continuously applied to a digital signal analysis device. The analysis portion of the speech coder extracts the amplitude, pitch, and other key parameters of the signal and then looks up related values in the code book for the portion of sound it has analyzed. Only key parameters and code book values are transmitted. This results in data compression ratios of 4:1 to over 16:1.
Error Detection and Error Correction
To help reduce the effects of errors on data transmission, error detection, and error protection systems are used in most communication systems. Error detection systems use a process of adding some data bits to the transmitted data signal that are used to help determine if bits were received in error due to distorted transmission. Error correction is made possible by sending bits that have a relationship to the data that is contained in the desired data block or message. These related bits permit a receiver of information to use these extra information bits to detect and/or correct for errors that may have occurred during data transmission.
A common measurement of the performance of a communication system is the amount of bits received in error, called the bit error rate (BER). The BER is the ratio of bits received in error compared to the total number of bits received.
Error detection processing involves the creation of additional bits that are sent with the original data. The additional check bits are created by using a formula calculation on the digital signal prior to sending the data. After the digital signal is received, the formula can be used again to create check bits from the received digital signal. If the check bits match, the original digital signal was received correctly. If the check bits do not match, some (or all) of the digital signal was received in error. This process is called error detection.
Some digital systems use sophisticated mathematical formulas to create the check bits so that the check bits can be used to make corrections (or predictions of the correct bits) to the received digital signal. This process is called error correction.
Figure below shows the basic error detection and correction process. This diagram shows that a sequence of digital bits is supplied to a computing device that produces a check bit sequence. The check bit sequence is sent in addition to the original digital bits. When the check bits are received, the same formula is used to check to see if any of the bits received were in error.
Echo Cancellation
Echo cancellation is a process of extracting a delayed version of an original transmitted signal (audio echo) from the received signal. Echoes may be created through acoustic feedback where some of the audio signal transferring from a speaker into a microphone.
Echoed signals cause distortion and may be removed by performing via advanced signal analysis and filtering. Figure 3.18 shows an example of the echo cancellation process. This diagram shows how the combining of two signals, the original plus a delayed version of the original produces a complex signal. The echo canceling system analyzes the complex signal and uses the signal analysis to create variations of the likely echo signal. This prediction of echoed signal is subtracted from the complex signal to reproduce the original signal without the echo.
Echoed signals can also occur in signals other than audio signals. When echoes occur on radio channels (the broadband signal), it is usually the result of the same signal that travels on different paths to reach its destination. This is called multipath propagation. Echo canceling can be used to reduce the effects of radio multipath propagation.
Digital Signal Processor (DSP)
A digital signal processor (DSP) is an electronics device or assembly (typically an integrated circuit) that is designed to process signals through the use of embedded microprocessor instructions. The use of DSPs in communication circuits allows manufacturers to quickly and reliably develop advanced communications systems through the use of software programs. The software programs (often called modules) perform advanced signal processing functions that previously complex dedicated electronics circuits. Although manufacturers may develop their own software modules, DSP software modules are often developed by other companies that specialize in specific types of communication technologies. For example, a manufacturer may purchase a software module for echo canceling from one DSP software module developer and a modulator software module from a different DSP software module developer. Because DSPs use these software modules, if new technologies such as speech compression, channel coding, or modulation techniques are developed, the manufacturer only has to change the software programs in the DSP to utilize the new technology.
Figure below shows typical digital signal processor that is used in a digital communication system. This diagram shows that a DSP contains a signal input and output lines, a microprocessor assembly, interrupt lines from assemblies that may require processing, and software program instructions. This diagram shows that this DSP has 3 software programs, digital signal compression, channel coding, and modulation coding. The digital signal compression software analyzes the digital audio signal and compresses the information to a lower data transmission rate. The channel coding adds control signals and error protection bits. The modulation coding formats (shapes) the output signal so it can be directly applied to an RF modulator assembly. This diagram also shows that an optional interface is included to allow updating of the software programs that are stored in the DSP.
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