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ONLINE COURSE 2201 THE PSTN
LESSON L1004 THE VOICEBAND

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The Voiceband

This free lesson is part of Teracom's Online Course "POTS and the PSTN".

POTS and The PSTN is the first course in the CTNS Certification Package, a set of six courses plus Certified Telecommunications Network Specialist (CTNS) certification from the Telecommunications Certification Organization (TCO).

In this lesson, we examine the bandwidth provided with Plain Ordinary Telephone Service, which is often referred to as the voiceband.  We’ll understand what the term “bandwidth” means, and how it is measured in the analog world.  We’ll look at the details of the voiceband, what frequencies it covers and why, and its limitations.

 
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Lesson Notes
If a tree falls in the forest and no-one is there to hear it, does it cause a sound?  The answer depends on whether you believe sound is sound pressure waves – air molecules getting closer together then further apart; or if you believe sound is the sensation one gets in one’s brain when one hears the sound pressure waves.

Then, two points of view when designing the telephone system would be:

  • At the receiver, reproduce the sound pressure waves exactly the same as they are at the transmitter, or
  • Reproduce the sensations in the listener’s brain just as if they had been speaking directly to the talker.

The difference between these two ideas is that the brain is a hugely complicated processing instrument, and we can play different stimuli at it and get the same response.

 
the voiceband
Voiceband Filter

What choice did Alexander Graham Bell make? 

Answer #2.  Based on testing human beings’ ears, throats and brains, and some technical limitations, A. G. Bell decided to transmit all of the information in the frequency range between about 300 and 3300 Hz.  Hertz (Hz) is the unit for frequency, or changes per second. 

This range or band of frequencies is called the voiceband.

The image is a representation of the voiceband, with frequency on the horizontal axis and amplitude or intensity on the vertical axis.

A simplified version of the image (contained within the lesson) shows any electricity vibrating at least 300 times per second and less often than 3300 times per second will be passed.

Any electricity vibrating less often than 300 times per second will be suppressed.  Similarly, any electricity vibrating more than 3300 times per second will be suppressed.  Only electricity vibrating within the band 300-3300 Hz will be transmitted.

The suppression of energy outside this frequency band 300-3300 Hz is implemented with simple electrical circuits called filters.  There is a filter in the telephone and a filter in the switch in the CO.

For our purposes, the term bandwidth means capacity.  In the analog world, capacity is measured literally by the width of the available frequency band.  In this case, the width of this frequency band is 3300 – 300 = 3000 Hz, or 3 kHz for short.

This 3 kHz bandwidth is the capacity provided for ordinary telephone service.

 

Why does the voiceband stop at 3300 Hz?

The two wires that make up the loop are capable of supporting electricity vibrating more often than 3300 times per second – in fact, DSL technologies require electricity vibrating at frequencies measured in the millions of times per second.

The users’ ears and brains are capable of detecting sound pressure waves vibrating more often than 3300 times per second – the human hearing range is traditionally thought to extend up to 20,000 Hz.

So why would the capacity a user is allowed to employ purposely limited to 3 kHz, even though the wires are capable of more than that, and the users are capable of more than that?

The answer is, as usual, money.  The more capacity a user is allowed to employ on the access circuit, the loop, the more capacity and hence money required to transmit the information across distance.

This narrow voiceband frequency range was chosen based on studying people’s ears, throats and brains, to determine the minimum capacity necessary to meet the requirements.

Returning to the question of trees falling in the forest: the sound pressure waves at the far end are not reproduced exactly as they were at the near end; in fact, they are quite muffled and distorted, missing most of the higher frequencies. 

The sound is reproduced just well enough so that the listener can recognize the speaker and understand what the speaker is saying, thus meeting the requirement to communicate information using speech and hearing.

We are interested in transmitting the minimum required to meet that objective since there is a direct relationship between the capacity a user can employ on the access circuit and the cost of transmitting the information long-distance.

 
It turns out that the voiceband is not quite enough

It turns out that the voiceband is not quite enough bandwidth to be able to understand everything the speaker is saying! 

In particular, it is difficult to tell the difference between “S” and “F” over a telephone.  This is because the frequency of sound pressure wave that distinguishes “S” from “F” is above 3300 Hz… which is not transmitted over the phone system.  Thus, one has to say “S as in Sierra” and “F as in Foxtrot”.

One could also say things like “S as in Sea”, “A as in Are” and “E as in Eye” to liven things up. 

If that doesn’t get the listener confused, there’s always “E as in Ewe”.
 
Summary
In this lesson, we have examined the capacity provided with Plain Ordinary Telephone Service, which is the frequency band from 300 – 3300 Hz, known as the voiceband.
We understood:
  • Why it is possible to restrict the capacity – the question of reproducing sound vs. reproducing thoughts in a person’s brain
  • Why the capacity is restricted – to reduce transmission costs
  • How the capacity is restricted – filters in the telephone and line card
  • Limitations of the voiceband, particularly with the sounds “s” and “f”.
 
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Overview of Courses included in the CTNS Certification Package
Like Teracom's famous core training Course 101 "Telecom, Datacom and Networking for Non-Engineers", our very popular core training DVD-Video packages and the Telecom 101 textbook, the Certified Telecommunications Network Specialist Certification Package begins with the Public Switched Telephone Network, then a course on Wireless Telecommunications, followed by four courses covering IP telecommunications and IP telecom networks.
If you are interested only in IP telecommunications, the CIPTS: Certified IP Telecom Network Specialist package may be appropriate, as it skips the traditional telephony and wireless and goes directly to the IP telecommunications courses.
If your goal is to build a full, rounded knowledge of telecommunications, then understanding the history, structure and operation of the telephone network built over the past 135 years or more is the starting point for everything else.
 
The CTNS package begins with Course 2201 POTS and the PSTN
On completion of this course, you will be able to draw a model of the Public Switched Telephone Network, identify its components and technologies including loops and trunks, switches, circuit-switching, explain the characteristics and operation of Plain Ordinary Telephone Service, analog and the voiceband, the outside plant, fiber to the neighborhood and remotes.
We begin with a history lesson, understanding how and why telephone networks and the companies that provide them are organized into local access and inter-city transmission, or as we will see, Local Exchange Carriers (LECs) and Inter-Exchange Carriers (IXCs).
Then we will establish a basic model for the PSTN and understand its main components: Customer Premise, Central Office, loop, trunk, outside plant, circuit switching, attenuation, loop length, remotes, and why knowledge of the characteristics of the loop remains essential knowledge even though we are moving to Voice over IP.
Next, we'll cover aspects of telephony and Plain Ordinary Telephone Service, including analog, the voiceband, twisted pair, supervision and signaling including DTMF. The course is completed with an overview of SS7, the control system for the telephone network in the US and Canada.
The next course in the CTNS package is Course 2206 Wireless Telecommunications
On completion of this course, you will be able to draw a model of a cellular wireless telecommunications network, identify all of its components and technologies from handset to mobile switch, and explain the characteristics and operation of FDMA, TDMA, CDMA and OFDMA, and how each was deployed for AMPS, GSM, UMTS and LTE. You will also be able to explain WiFi, the frequencies used and the different 802.11 standards, and the similarities and differences between LEO and GEO satellites.
In many parts of the world, particularly outside Canada, the US and Western Europe, this is the physical telephone network, as deploying radio transceivers is far cheaper than embarking on a new project to pull copper wires and/or fiber to every residence.
Most of this course is devoted to mobile wireless telecommunications. We begin with basic concepts and terminology including base stations and transceivers, mobile switches and backhaul, handoffs, cellular radio concepts and digital radio concepts.
Then, we cover spectrum-sharing technologies and their variations in chronological order: GSM/TDMA vs. CDMA for second generation, 1X vs. UMTS CDMA for third generation along with their data-optimized 1XEV-DO and HSPA, how Steve Jobs ended the standards wars with the iPhone and explaining the OFDM spectrum-sharing method of LTE for 4G.
This course is completed with a lesson on WiFi, or more precisely, 802.11 wireless LANs, and a lesson on satellite communications.
The remaining four courses in the CTNS package are on the "IP" telecommunications network and its three main enabling technologies: Ethernet, IP and MPLS, and beginning with the OSI model and its layers to establish a framework.
If you'd prefer to take just these four "IP" courses, please check out the Certified IP Telecom Network Specialist package.
Course 2212 The OSI Layers and Protocol Stacks
On completion of this course, you will be able to define a protocol and differentiate that from a standard, explain why a layered architecture is required, list the seven layers of the OSI model, the name, purpose and functions of each one, and explain how a protocol stack operates and where the protocol headers are located.
This course establishes a framework for all of the subsequent discussions: the OSI 7-Layer Reference Model, which identifies and divides the functions to be performed into groups called layers. This framework is required to sort out the many functions that need to be performed, and to be able to discuss separate issues separately.
First, we'll define the term "protocol" and compare that to a standard. Then we'll define "layer" and how a layered architecture operates, and provide an overview of the name, purpose and function of each of the seven layers in the OSI model.
Then, we'll go back through the story more slowly, with one lesson for each of the layers, examining in greater detail the functions that have to be performed and giving examples of protocols and how and where they are used to implement particular layers.
The result is a protocol stack, one protocol on top of another on top of another to fulfill all of the required functions. To make this more understandable, this course ends with the famous FedEx Analogy illustrating the concepts using company-to-company communications, and an analogy of Babushka dolls to illustrate how the protocol headers are nested at the bits level.
Next is Course 2211 Ethernet, LANs and VLANs – which could also be titled "Layer 2"
On completion of this course, you will be able to define the purpose and basic functioning of a LAN, what a broadcast domain is, explain Ethernet and the 802 standards from 10BASE-5 to 1000BASE-T and Gigabit Ethernet, MAC addresses, cable categories, bridging, how LAN switches work and what they do, VLANs, LAN jargon and buzzwords, and how LANs implement Layer 2 of the OSI model, moving data between two devices on the same circuit - or to be more precise, between two devices that are in the same broadcast domain.
As we will have established in the previous course, Layer 2 is all about communications between two devices that are on the same circuit, or more precisely, in the same broadcast domain. It turns out that this is implemented by moving frames with link addresses over physical connections following the 802 series of standards, colloquially referred to as Ethernet, MAC frames and MAC addresses.
We'll begin with the original LAN: Ethernet and its bus topology, defining "broadcast domain" and explaining its fundamental operation and characteristics: CSMA-CD access control, MAC addresses and MAC frames.
Then we'll cover the IEEE 802 standards and the evolution of Ethernet from 10BASE-T to Gig-E, LAN cables and the TIA-568 cable categories, basic cabling design; what "bridging" means and how a LAN switch works.
This course is completed with the important concept of VLANs: defining broadcast domains in software, a key part of basic network security practice.
Then Course 2213 IP Networks, Routers and Addresses – which could also be titled "Layer 3"
On completion of this course, you will be able to explain how IP packet networks implement Layer 3 of the OSI model, define bandwidth on demand and its advantages, what a router does, the basic structure of a routing table, where routers are located, define the IPv4 address structure and dotted-decimal notation, explain how both static and dynamic addresses are assigned using DHCP, what private addresses are and how they are interfaced to the public IP network, and the structure, allocation and assignment of IPv6 addresses.
This is a comprehensive course on Layer 3 of the OSI Model, concentrating on IP addresses, routers and packets. We begin with the two basic principles of packet networks: bandwidth on demand, also known as overbooking or statistical multiplexing; and packet-switching, also known as packet forwarding or routing.
We'll understand what routers do and where they are located, routing tables and the basic operation of a router and the standard strategy deploying an edge router between the LANs and the WAN at each location.
Then we'll cover IP version 4: address classes and how they are assigned to Regional Internet Registries then ISPs then end-users, dotted-decimal notation, static addresses, dynamic addresses and DHCP, public addresses, private addresses and NAT.
The course concludes with IPv6: the IPv6 packet and changes from IPv4, IPv6 address allocations and assignments and end up understanding how IPv6 subnets will be assigned to broadcast domains and 18 billion billion addresses per residence.
The last course in this certification package is Course 2214 MPLS and Carrier Networks
On completion of this course, you will be able to draw a model for a service provider's network, define the terms Customer Edge and Provider Edge, explain what a traffic profile is and how that relates to a Service Level Agreement, how Frame Relay got its name, what ATM is and why it is on the way out, the purpose, components, terminology and operation of MPLS, and how MPLS can be used to implement integration or convergence, aggregation and differentiated classes of service – what people mean when they say "MPLS service" and its pros and cons compared to Internet service.
This is an extensive and comprehensive course devoted to the structure, components and operation of carrier packet networks and services, how they are implemented, packaged and marketed, and how they are used by government, business and other carriers.
The IP packets and routing of the previous course is one part of the story. Performance guarantees, and methods for quality of service, traffic management, aggregation and integration is another big part of the story, particularly once we leave the lab and venture into the real world and the business of telecommunications services.
We'll begin by establishing a basic model for a customer obtaining service from a provider, defining Customer Edge, Provider Edge, access and core, and a Service Level Agreement: traffic profile vs. transmission characteristics.
Next, we'll understand virtual circuits, a powerful tool used for traffic management and variations like connection-oriented vs. connectionless communications and reliable vs. unreliable network services.
With the fundamentals in place, we will survey the different technologies used to implement this in practice: Frame Relay, ATM and MPLS, explaining the equipment, jargon and principles of operation, and the advantages each technology has over the previous. In particular, we'll understand the big advantage of MPLS over Frame Relay in the user-network interface.
Once we've covered all of the components of an MPLS network and its operation, we'll see how MPLS is used to implement Diff-Serv, i.e. different classes of service, how MPLS is used to implement integration or "convergence" of services onto a single network service, and how MPLS is used to aggregate traffic for management.
The course is completed with a lesson on "MPLS service", and how that compares to Internet service.
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The knowledge you gain taking Teracom's Online Courses, confirmed with TCO Certification, is foundational knowledge in telecommunications, IP, networking and wireless: fundamental concepts, mainstream technologies, jargon, buzzwords, and the underlying ideas - and how it all fits together.
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