Analog and Digital Communications Concepts



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Analog and Digital Communications Concepts




Representing data as Analog Signals

Converting Analog Data-to-Analog Signals

During the early development stages of copper-based analog telephone networks it was discovered that human voice could be carried adequately from 300Hz to 3330 Hz. As a result telephone networks were originally designed to transmit voice conversations within a range of 3000 Hz. Voice signals that generate a signal less than 300 Hz or greater than 3330 Hz are discarded.


There are 2 basic ways in which analog data are represented as analog signals:

  • At their original frequency, called a baseband signal

  • Or at a different frequency

When we pick up a telephone and speak into it, the telephone network transmits our analog voice signal at face value (somewhere between 300 and 3330 Hz). This face value is an example of a baseband signal.


Alternatively, the telephone network can combine our signal into another signal (called a carrier), and then transmit these combined signals at a different frequency.
A carrier is a continuous signal that operates at a predefined frequency. Changing a carrier so that it can represent data in a form suitable for transmission is called modulation.
Characteristics of a carrier that can be modified are Amplitude, Frequency and Phase.
Modifying the Amplitude of a wave is called Amplitude Modulation

(Changing the signal’s strength)
Modifying the Frequency of a wave is called Frequency Modulation

(Changing the signal’s pitch)
Modifying the Phase of a wave is called Phase Modulation

(Temporarily delaying the natural flow of the waveform)


Converting Digital Data-to-Analog signal

Transmitting digital data (output from a computer) across an analog based communication network, is done by modifying (modulating) a continuous signal (Carrier) at the sender end so that the signal conforms to the digital data being transmitted, and then converting the signal back into digital form at the receiver.


The device that performs these functions is called a modem, which is a contraction of the words modulator and demodulator. Two modems are required- one at each end of the transmission line- and both modems must use the same modulation technique. A sending modem first produces a continuous carrier signal and then modified the signal using a specific modulation technique. When the modified signal is received by the receiving modem, the signal is then converted back into digital form.
When used in the context of converting digital data into analog signals the modulations techniques are called:

  • Amplitude-shift keying (ASK)

  • Frequency-shift keying (FSK)

  • Phase-shift keying (PSK)


ASK this process involves varying the signal voltage while keeping the frequency constant. One amplitude is used to represent a binary 0 and a second amplitude is used to represent a binary 1. A sending ASK modem, generates a continuous carrier and used this unmodulated signal to represent a binary 0; the modem also modulates this signal (using ASK) to represent a binary 1.
FSK Alters the frequency (cycles per second) of a signal so that it conforms to the digital data. Amplitude is kept constant. One frequency is used to represent a binary 0 and a second frequency is used to represent a binary 1. A sending FSK modem generates a continuous carrier and let’s the signals standard frequency represent a binary 0. Whenever a 1 needs to be converted, the modem can either reduce or increase the frequency.
PSK modifies the phase angle of a carrier wave based on the digital data being transmitted. The changes in phase angle are what convey the data in a phase-modulated signal.

Changing the phase of a wave enables data encoding with more than one bit of data at a time. If a phase shift occurs at 0, 90, 180 and 270 0 then 2 bits of information (called dibits) can be transmitted for each signal change. These dibits pair 00, 01, 10, and 11 respectively. Similarly if phase shift occur at 8 different angles (0, 45, 90, 135, 180, 225, 270 and 3150 then these signal represent three bits of information (tribits). These tribits are respectively 000, 001, 010, 011, 100, 101, 110, and 111. As a result PSK can encode up to three bits per baud.


QAM Phase shift can also be combined with Amplitude modulation. One common strategy employed is called Quadrature Amplitude Modulation, which uses eight phase changes and two amplitudes to create 16 different signal changes. A QAM can encode between four and seven bits per baud. A modified version of QAM called trellis-coded modulation (TCM) incorporates extra bits for error-correction. Both QAM and TCM provide high data transfer rates because they are able to incorporate several bits per signal change.
Modem Demo

Representing Data as Digital Signals

As digital technology and computer data applications emerged, analog technology was unable to separate data from noise in a satisfactory manner. This led to the introduction of digital signalling, which requires converting analog signals to digital signals.



Converting Analog-Data-to-Digital signals

Representing Analog data as digital signals requires converting the data’s corresponding analog signal, which is in the form of a sine wave, into digital signal, which is represented by 0’s and 1’s The most common approach is a process known as pulse-code modulation (PCM), and involves two steps: Sampling and coding.


Digitizing an analog signal requires taking regular samples of the amplitude of the signal’s waveform over time so that the generated digital signal matches the corresponding analog signal. According to a sampling theorem known as Nyquist’s Rule, if an analog signal is sampled at regular intervals and at twice the highest frequency on the line, then the sample will be an exact representation of the original signal.
Even though voice transmission requires 3000 Hz bandwidth, phone companies allocate 4000 Hz and install filters at the 300 Hz and 3300 Hz. Therefore actual sampling is 8000 per second when converting voice to digital form.
Once a sample has been taken, it must be converted into binary digit where 0 and 1 represent the absence or presence of voltage, respectively. Determining whether a sample gets coded 0 or 1 depends on where the sample was taken.
If an eight-bit sample is used, then the sound wave can be partitioned into 256 (28) possible points. The first 128 points (0 to 127) get coded 00000000, and the last 128 (128 to 255) get coded 00000001. After the sampling and coding steps are complete, the resulting digital codes are then transmitted as a digital signal waveform.
Digitizing an analog signal via PCM is done using a device called a codec (coder-decoder), which can be thought of as the opposite of a modem. A codec converts analog data into a digital signal; a modem on the other hand converts digital data into an analog signal.


This is an example of what happens when a computer using a modem communicates with another computer equipped with a modem over the telephones lines.
The telephone company maintains analog lines from clients to the local switching stations. In between switching stations lines are digital. The signal from the computer at the sending end has to be converted from digital to analog, when the signal arrives at the switching station it is converted again from analog to digital. Again when the signal arrives at the second switching station is has to be converted again to analog…. Finally at the destination computer the signal is converted again from analog to digital… In total it required 4 conversions for these two computers to effectively communicate.
This is the role of a modem (modulator / demodulator) to convert a digital signal into an analog one at the sending end and to convert it from analog to digital at the receiving end.


Converting Digital data-to-Digital Signals

Transmitting digital data across a digital network (Ethernet LAN) requires representing the digital data as a digital signal. Three common techniques are used for this task are:



  • Manchester encoding

  • Differential Manchester coding

  • Non-Return to Zero, Invert on ones (NRZI)


Manchester encoding Used on Ethernet/802.3 networks. Its main benefit is error recovery. A 1 is sent as a half-time-period low followed by a half-time-period high, and a 0 is sent at half-time-period high followed by a half-time-period low. Consequently, the end of the last transmitted bit is easily determined immediately following the transmission of the last bit.
Differential Manchester Encoding used on token ring networks. Similar to Manchester encoding: each bit-period is partitioned into two intervals and a transition between high and low occurs during each bit period. The difference in between the two methods is in the interpretation of this transition. In Differential Manchester Encoding the interpretation of low-to-high or high to low is a function of the previous bit-period. More specifically, the presence of a transition at the beginning of a bit-period is coded 0, and the absence of a transition at the beginning of a bit-period is coded 1.
NRZI used on Fast Ethernet and FDDI (Fiber Distributed Data Interface). Instead of user level voltages to encode the data, encoding is based n transitions from one voltage state to another. Data are coded 0 if no transition occur, but are coded 1 at the beginning of a transition.
An application of NRZI is an encoding strategy known as 4B/5B (four-bits to five bits) method. The 4B/5B-encoding scheme takes data in four-bit codes and maps it to corresponding five-bits codes. By transmitting five-bits codes using NZRI, a logical 1-bit is transmitted at least once every five sequential data bits resulting in a signal transmission. This 4B/5B scheme makes it possible for a LAN to operate at 125MHz and provides a data rate of 100Mb/sec. Note that the use of one extra bit for every five translate to 20% overhead for every clock encoding. In contrast, Manchester Encoding requires 50% bandwidth overhead for clock encoding because it guarantees at least one signal transition for every bit transmitted.

Digital Carrier Systems

T1 and DS circuits

Digital signalling and the T-carrier system were introduced to resolve attenuation and noise amplification.



  • Repeaters are being used to regenerate signals

  • T-carrier system, which uses TDM (Time Division Multiplexing) to support multiple channels in a single digital signal, was the first systems designed to implement digitized voice transmission.



T1, a product of T-carrier, describes the multiplexing of 24 separate voice channels into one single wideband digital signal.

  • A T1 frame consists of 193 bits – eight bits per channel and one bit plus one bit for framing. (168 for data, 24 for control and 1 for synchronization)

  • Each voice channel is digitized using PCM (pulse-code modulation) and has a data rate of 64 kbps

  • When multiplexed into a digital signal a voice channel is referred to as a Digital signal at level 0 (DS-0), thus DS-0 has a data rate of 64 kbps.

  • A T1 circuit carries a DS-1 signal, which consists of 24 DS-0 channels plus one 8 kbps channel reserved for framing. This results in an aggregate bandwidth of 1.544 Mbps.

    • Data 56,000 bps per channel at 24 channels = 1,344,000 bps

    • Control 8,000 bps per channel at 24 channels = 192,000 bps

    • Framing 8,000 bps for frame synchronization = 8,000 bps

  • Two T1 lines are combined to from a T1C circuit rated at 3.152 Mbps

  • A T2 circuit (DS-2) consists of 4 multiplexed T1 circuits and has an aggregate bandwidth of 6.312 Mbps

  • A T3 link (DS-3) consists of 28 multiplexed T1 circuits with an aggregate bandwidth of 44.736 Mbps

  • A T4 channel (DS-4), rated at 274.176 Mbps, consists of 168 multiplexed T1 circuits.




NADH Lines have the same meaning in Australia, Japan as it does in North America. However in Europe, South America, Africa, part of Asia and Mexico, an analogous service called E-1 is used in these locations. An E-1 carrier supports thirty 64 kbps channels plus 2 signalling and control channels. E-1 links can be multiplexed into higher capacity lines.

A T1 circuit requires special termination equipment called CSU/DSU



  • A Channel Service Unit (CSU) regenerates the signal, monitors the line for electrical anomalies, provides proper electrical termination, performs framing, and provides remote loopback testing for diagnosing line problems.

  • A Data (or Digital) Service Unit (DSU) provides the interface to connect a remote bridge, router, or switch. Also provide flow control between the network and the CSU.

  • An E-1 circuit is terminated using a Network Termination Unit (NTU), which provides broadly similar CSU/DSU functionality.






Document on CSU/DSU

Fractional T1 (FT1)





  • Provides a fraction of a T1 capacity.

  • Achieved by combining multiple DS-0 channels

    • 128 kbps => 2 DS-0 channels

    • 256 kbps => 4 DS-0 channels

    • 512 kbps => 8 DS-0 channels

  • When ordering FT1 you actually get a full T1 channel, but only pay for the number of DS-0 channels you use.

  • Less efficient than ISDN (Integrated Services Data Networks, Chapter 12), in terms of bandwidth available for data communication. In FT1 controls are inband, meaning that the 8 kbps required for control are amongst the 64 kbps bandwidth. On the other hand on ISDN separate channels are used for control and therefore provides a full 64kbps bandwidth.

  • FT1 service is attractive to customers who do not require a full T1 service but need more capacity than an ISDN (64/128-kbps) line.



SONET and OC Circuits



SONET (Synchronous Optical NETworks) and SDH (Synchronous Digital Hierarchy) are both International standards that provide specification for high-speed digital transmission via optical fibre. This involves converting signals in electrical to optical form at the source and an optical-to-electrical at the destination.


  • SONET Developed by ANSI

  • SDH developed by ITU-T, drafted after SONET and incorporates it

  • Both standards work at the physical layer of the OSI model

  • The building block of the SONET signal hierarchy is STS-1 (51,84 Mbps) The line rate is derived from the STS-1 frame and consists of 810 eight-bit bytes transmitted at 8000 Hz.

  • In the ANSI world, SONET’s terminology include Optical Carrier level (OC-n) and signals over copper Synchronous transport Signal (STS-n)

  • In the ITU-T world, the official term used is Synchronous Transport Module (STM-n).

Advantages over copper based (T1) hierarchy



  • Hundreds of thousands of simultaneous voice and data transmission are possible

  • Immune to EMI (Electro Magnetic Interference)

  • Fibre is available in either single mode or multimode, therefore making it suitable for being used for LAN connections or as the backbone for a WAN.

  • Bandwidth can be allocated on an as-needed basis, routes can be dynamically reconfigured

  • Can serve as the transport facility for any type of network technology or service, including ATM (Asynchronous Transport Mode), FDDI (Fiber Distributed Data Interface), SMDS (Switched Multimegabit Data Service) and ISDN (Integrated Services Digital Network).

  • Can support various topologies including point-to-point, star, and ring.


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