TELECOMMUNICATION SYSTEMS
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1 TELECOMMUNICATION SYSTEMS By Syed Bakhtawar Shah Abid Lecturer in Computer Science 1
2 MULTIPLEXING An efficient system maximizes the utilization of all resources. Bandwidth is one of the most precious resources we have in data communications. A medium can natively carry only one signal at any moment in time. For multiple signals to share a medium, the medium must somehow be divided, giving each signal a portion of the total bandwidth 2
3 MULTIPLEXING 3 Whenever the bandwidth of a medium linking two devices is greater than the bandwidth needs of the devices, the link can be shared. Multiplexing is the set of techniques that allows the simultaneous transmission of multiple signals across a single data link.
4 MULTIPLEXING In a multiplexed system, n lines share the bandwidth of one link. 4
5 MULTIPLEXING There are three basic multiplexing techniques: Frequency-division multiplexing, Wavelength-division multiplexing, and Time-division multiplexing. The first two are techniques designed for analog signals, the third, for digital signals. 5
6 MULTIPLEXING 6
7 Time-Division Multiplexing Time Division Multiplexing is further divided into two broad categories Synchronous Time-Division Multiplexing Statistical Time-Division Multiplexing 7
8 Synchronous Time-Division Multiplexing 8 Time-division multiplexing (TDM) is a digital process that allows several connections to share the high bandwidth of a line. Instead of sharing a portion of the bandwidth as in FDM, time is shared. Each connection occupies a portion of time in the link.
9 Synchronous Time-Division Multiplexing Information from channel 1 is sampled and transmitted first, information from channel 2 is then sampled and transmitted, and so on in a regular sequence, cycling back to channel 1 and continuing. Bandwidth allocation is static, in the sense that each channel has a fixed, predetermined bandwidth. Because TDM is protocol insensitive, it is capable of combining various higher-layer protocols onto a single high-speed transmission link. 9
10 Time-Division Multiplexing If an end-user device does not have data to transmit, empty time slots are transmitted. Such a TDM mechanism is also known as synchronous time-division multiplexing. 1 0 We can see this in Figure where device B & C is not transmitting data. Statistical multiplexing overcomes this inefficiency by only transmitting data from active end devices.
11 Data Rate Management One problem with TDM is how to handle a disparity in the input data rates. if data rates are not the same, the following three strategies, or a combination of them, can be used. Multilevel multiplexing, Multiple-slot allocation, and Pulse stuffing. 11
12 MULTILEVEL MULTIPLEXING 12
13 MULTIPLE-SLOT ALLOCATION 13
14 PULSE STUFFING Pulse stuffing is also called bit padding, or bit stuffing 14
15 FRAME SYNCHRONIZING If the multiplexer and the de-multiplexer are not synchronized, a bit belonging to one channel may be received by the wrong channel. One or more synchronization bits are usually added to the beginning of each frame. These bits, called framing bits, follow a pattern, frame to frame, that allows the de-multiplexer to synchronize with the incoming stream so that it can separate the time slots accurately. In most cases, this synchronization information consists of 1 bit per frame, alternating between 0 and I 15
16 Statistical Time-Division Multiplexing If an end-user device is not active, no space is wasted on the multiplexed stream. A statistical multiplexer accepts the incoming data streams and creates a frame containing only the data to be transmitted. In synchronous TDM, each signal is given a unique but equal time slot for its information that is interleaved with the others. In statistical TDM (also known as asynchronous TDM), however, the amount of time per slot is variable. 1 6
17 Statistical Time-Division Multiplexing 17 In statistical time-division multiplexing, slots are dynamically allocated to improve bandwidth efficiency. Only when an input line has a slot's worth of data to send is given a slot in the output frame. In statistical multiplexing, the number of slots in each frame is less than the number of input lines.
18 Statistical Time-Division Multiplexing The multiplexer checks each input line in round robin fashion; it allocates a slot for an input line if the line has data to send; otherwise, it skips the line and checks the next line. 18
19 Analog Signal Processing Analog signal refers to the content of a transmission being determined by the strength, amplitude or frequency of a signal. Analog signal has infinitely many levels of intensity over a period of time. 1 9 In the case of telephony, for instance, When you speak into a handset, there are changes in the air pressure around your mouth. Those changes in air pressure fall onto the handset, where they are amplified and then converted into current, or voltage fluctuations. Those fluctuations in current are an analog of the actual voice pattern hence the use of the term analog to describe these signals.
20 Analog Signal Processing 20 The human voice can typically generate frequencies from 100Hz to 10,000Hz, for a bandwidth of 9,900Hz. But the ear does not require a vast range of frequencies to elicit meaning from ordinary speech; the vast majority of sounds we make that constitute intelligible speech fall between 200Hz and 3,400Hz. Analog telephony signals span the 200-Hz to 3.4-KHz frequency band. Such analog signals are referred to as narrowband due to their narrow frequency response. The phone company typically allot a total bandwidth of 4,000Hz or 4KHz or less for voice transmission. Analog video signals operate in a frequency band from (0 Hz) up to 60 MHz. Such analog signals are referred to as broadband due to their wide frequency response.
21 Analog-To-Digital Conversion Converting an analog telephony signal to a digital signal involves Filtering, Sampling, Quantization and Encoding 21
22 Filtering Audio frequencies ranges from 20 Hz to 20,000 Hz. Telephone transmission systems are designed to transmit analog signals between 200 Hz and 3400 Hz. End frequencies below 200 Hz and above 3400 Hz are removed by a process called filtering or Band Pass Filtering (BPF). 22 BPFs are constructed using analog electronic components, such as capacitors and inductors. As indicated in the Figure above, a band pass filter (BPF) is used to filter the audio telephony band for analog-to-digital (A/D) conversion.
23 SAMPLING / PAM In the sampling process, portions of a signal are used to represent the whole signal. Each time the signal is sampled, a PAM signal is generated. According to the Nyquist theorem, to accurately reproduce the analog signal (speech), a sampling rate of at least twice the highest frequency to be reproduced is required. Because the majority of telephony voice frequencies (200 to 3400 Hz) are less than 4 khz, an 8-kHz sampling rate has been established as the standard. 23 The PAM sampler measures the filtered analog signal 8000 times per second, or once every 125 microseconds.
24 Sampling V.90 or V.92 Standard modems are available with a bit rate of 56,000 bps or 56 kbps. Also called 56K Modem. The telephone companies sample 8000 times per second with 8-bits per sample. One of the bits of each sample is used for control purposes, which means each sample is 7-bits. The rate is therefore 8000x7, or 56,000 or 56kbps. Each colored TV channel required minimum bandwidth of 6MHz. 24
25 Quantization 25
26 Quantization 26
27 Quantization 27
28 Encoding (Pulse Code Modulation (PCM)) The decimal (base 10) number derived via quantization is then converted to its equivalent 8-bit binary number. As illustrated in Figure, the output is an 8-bit "word" in which each bit can be either a 1 (pulse) or a 0 (no pulse). 28 This process is repeated 8000 times a second for a telephony voice channel service. The output (8000 samples/second * 8 bits/sample) is a 64-kbps PCM signal. This 64-kbps channel is called a DS0, which forms the fundamental building block of the digital signal level (DS level) hierarchy.
29 Pulse Code Modulation (PCM) Its is the most common technique to change an analog signal to digital data (digitization). A PCM encoder has three processes, 29
30 Full PCM Process 30
31 Key terms & Concepts 31 Bandwidth The range of frequencies (that is, the difference between the lowest and highest frequencies carried) that make up a signal is called bandwidth. There are three major classes of bandwidth that we refer to in telecommunications networks: Narrowband Narrowband means that you can accommodate up to 64Kbps, which is also known as the DS-0 (Digital Signal level 0) channel. This is the fundamental increment on which digital networks were built. If we combine these 64Kbps channels together, we can achieve wideband transmission rates. Broadband ITU has defined broadband as being anything over 2Mbps. This definition was created in the 1970s, when 2Mbps seemed like a remarkable capacity.
32 Key terms & Concepts 32 Wideband Wideband is defined as being n x 64Kbps, up to approximately 45Mbps. A range of services are provisioned to support wideband capabilities, including T-carrier, E-carrier, and J-carrier services. These are the services on which the first generation of digital hierarchy was built. T-1 offers 1.544Mbps. T-carrier system is an North American standard, T-carrier is used in the United States, Canada, Korea, Hong Kong, and Taiwan. E-1, which provides a total of 2.048Mbps, is specified by the ITU. It is the international standard used throughout Europe, Africa, most of Asia- Pacific, the Middle East, and Latin America. J-carrier is the Japanese standard, and J-1 offers 1.544Mbps.
33 Digital Signal Services Telephone company implement TDM through a hierarchy of digital signal, called Digital Signal (DS) Services or digital hierarchy. The following figure shows the data rate supported by each level. 33 TDM circuits typically use multiplexers, such as channel service units/digital service units (CSUs/DSUs) or channel banks at the CPE (customer premises equipment) side, and they use larger programmable multiplexers. The T-carrier system is entirely digital, using PCM and TDM.
34 Digital Signal Services A DS-0 service is a signal digital channel of 64 kbps. DS-1 is a Mbps service. It can be used as a single service for Mbps or it can be used to multiplex 24 DS-0 channels or a combination of these services types. DS-2 is a Mbps service. It can be used as single service for Mbps transmission, or It can be used to multiplex 4 DS-1 channels, 96 Ds- 0 channels or a combination of these services types. DS-3 is Mbps service. It can be used as a single service for Mbps transmission or It can be used to multiplex 7 DS-2 channels, 28 DS-1 channels, 672 DS-0 channels or a combination of these services types. DS-4 is a Mbps service. It can be used as a single service for Mbps transmission or It can be used to multiplex 6 DS-3 channels, 42 DS-2 channels, 168 DS-1 channels, 4032 DS-0 channels or a combination of these services types. 34
35 The T-Carrier DS-0, DS-1, and so on are the name of the services. To implement those services, the telephone companies use T- Carrier or Line (T-1 to T4). These are carriers or Lines with the capacity precisely matched to the data rates of the DS-1 to DS-4 services. 35
36 The T-1 Frame Structure The frame used on a T-1 line is usually 193-bits divided into 24 slots of 8-bit each plus 1 extra bit for synchronization (24x8+1=193); 36 Each slot contain one signal segment for each channel; 24 segments are interleaved in one frame. If a T-1 carries 8000 frames, the data rate is Mbps (193x8000)=(1.544 Mbps)- the capacity of the line.
37 The E-Carrier Europeans use a version of T-carriers called E-carriers. The two systems are conceptually identical, but their capacity differ. Line Rate (Mbps) Voice Channels E E E E
38 The E-1 Frame Structure The E1 consists of 32 DS 0 channels. The E1 signal format carries data at a rate of Mbps. A Mbps basic frame is comprised of 256 bits numbered from 1 to 256. These bits provide 32 8-bit time slots numbered from 0 to 31. i.e. 32x8= 256x8000 = Mbps 38
39 µ-law and A-Law Coding The dynamic range is the difference in decibels (db) between weaker (softer) and stronger (louder) signals. The µ-law and A-law algorithms are standard compression algorithms used in digital communications systems to optimize and modify the dynamic range of an analog signal for digitizing. The µ-law is typically used on T1 facilities (American), whereas the A- law is used on E1 facilities (Europe). Companding is a method commonly used in telephony applications. Basically, the voice is sampled at 8000 samples per second and converted into a 14-bit word (µ-law) or 13-bit word (A-law) that goes into the compander. In companding (compression and expansion) the samples are processed using a nonlinear formula to transform them into 8-bit words. Nonlinear coding uses more values to represent lower-volume levels and fewer values for higher-volume levels (Huffman coding algorithm).
40 ? 40
41 Thank You 41
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