(W47M00) Introduction to POTS

Introduction to POTS

To understand the fundamentals and basic structure of VoIP (Voice over Internet Protocol), it's helpful to know how the first phone call was generated and how the POTS - Plain Old Telephone System evolved into digital systems.

The POTS is the traditional telephone system that initially used copper lines to carry analog voice from one geographic point to the other, usually from the customer end to the central telephone exchange (or central office (CO)). The copper cable connection from the telephone handset installed at the customer’s premises up to the central exchange is called local loop. It is also known as fixedline or landline.

Figure 1 – The POTS architecture

Communication has evolved from prehistoric times through several methods including:

  • Visual and Symbolic cave paintings
  • Written Tradition - development of alphabets
  • Use of homing pigeons to carry messages
  • Postal Services - up to the development of the POTS –  use of pure copper wires for transmission of voice signals without a switching circuit.

At this point, the concept of dialing a number to reach a destination still didn’t exist. All that existed was a switchboard controlled by an operator at the telephone exchange and the overheard copper pairs connection from the exchange right down to the customer’s residence.

The switchboard had, for every pair of wires entering the exchange, a 2 pin connection socket through which calls were patched to complete a call setup.

When a user wanted to place a call to another party, the handset would just be lifted - sending a signal to the exchange operator. The operator asks for the number or name of the other party. To connect this call, the operator connects a patch cord - a two wire cable with a jack plug on each end - to bridge the 2-pin connection sockets, one pair for the calling party and the other pair for the called party. Immediately after patching, the receiving party’s telephone handset would ring and the two parties could communicate. The established call terminates when either party hangs up the telephone. This type of call setup is called ringdown, a typical example of circuit switching – as the route between source and destination had to be established before message transmission. This mode of communication was made up of only analog voice signals being carried through the copper medium.

POTS ringdown Call flow:

  1. A calling party goes off-hook – a term used to indicate the state of a telephone handset being lifted from its base
  2. A signal (light above each socket of a copper pair) alerts the operator at the telephone exchange
  3. The operator talks to the calling party and requests the name of the other party to receive call
  4. Operator patches socket of calling party to socket of remote party to establish call
  5. Conversation between parties begins and the physical circuit is closed and blocked from all incoming calls until the call  terminates. The physical circuit is still closed even when no voice is transmitted from one party to the other - thus wasting the circuit’s bandwidth
  6. Call termination occurs when either of the two parties cuts the call and the light above either socket for the parties goes off

The major concerns for the POTS ringdown call setup were:

a.) The degradation of quality of voice due to noise on the copper pair

b.) Inefficient use of the bandwidth available on the copper pair

c.) The presence of an operator – who was in the position to listen to private conversations between parties

d.) The telephone exchange was a single point of failure. Immediately after the exchange goes down, no communication would be possible

Later, automated analog systems were developed to ensure the privacy of customers by eliminating the need for a telephone operator. Noise and bandwidth issues were taken care of when the POTS evolved into digital systems. After its evolution, the POTS became commonly known as Public Switched Telephone Network – PSTN.

Enhancements POTS acquired as it evolved into a Digital System

Call waiting – a feature allowing a POTS subscriber to accept an incoming call whilst placing an active call on hold. This feature was provided at an additional cost.

Call forwarding – a feature allowing the transfer of an incoming call to another phone at an added cost.

Caller Line Identification Presentation (CLIP) – This feature shows you the number of the person calling so you can decide whether to answer the call. To use you will need a compatible handset.

E.164 Numbering Plan – Just like a public IP address in computer networks, E.164 is the international telephone numbering plan that ensures each device on the POTS has a globally unique number. This is what allows phone calls to be correctly routed to individual phones in different countries.

In this plan, numbers are formatted:

[+] [country code] [subscriber number including area code] and can have a maximum of fifteen digits.

+48      22                           xxx xx xx                                                                                   Warsaw fixedline in Poland

+48      50                           x xxx  xxx                                                                                  Tmobile mobile network in Poland

+233    24                           xxx  xx  xx                                                                                 MTN mobile network in Ghana

+233    30                           2xx xx xx                                                                                   Accra fixedline in Ghana

Signaling – This is the exchange of controls required to setup a call, manage a call and then teardown a call.


Why is digitization required for POTS?

One of the main challenges observed when transmitting analog signals over a distance is that the signal suffers interference, which can cause low volume, static, etc. Instead of maintaining the analog signal over distances that may span several thousands of metres, certain characteristics of the original voice signal is measured and then sent to the far end or receiving end. In this method, even though the full original voice is not transmitted, all the information needed to reconstruct the signal is available. This method is known as Pulse Code Modulation (PCM).


How Pulse Code Modulation (PCM) changes pure analog voice signals in POTS into digital signals.

With the introduction of digital systems in POTS, the pure analog voice signals transmitted from sender to receiver (or between users) are sampled, quantized, and encoded before being transmitted over the copper medium, and the process is reversed for the recovery of voice signals.

Sampling: Because voice signals are analog in nature, they are continuous in both time and amplitude. In contrast, digital signals are individually separate and distinct. Sampling converts the analog signal from continuous time to discrete time. This is done by measuring the value of the signal at certain time intervals.

Quantizing:  After analog signals are measured at certain interval times, the continuous range of amplitude obtained are broken into a finite number of discrete levels or steps.

Encoding: This involves the assigning of binary codes to the output from quantizing.


Evolved POTS Call flow

Let’s now analyze the call flow for POTS after its evolution into a digital system. The call flow will start from a caller (party A) picking up the handset and attempting to place or originate a call to another caller (party B).

The flow is broken down into:

  1. off-hook (a state of telephone handset when it is lifted or separated  from its base or the general description of a user picking up the handset)
  2. digit-receipt
  3. ringdown
  4. conversation and
  5. on-hook sections

In this scenario, both parties (A & B) are connected through different exchanges or central offices (COs).

The following list outlines, in order, the actions performed by the network:


Party A picks up the phone, and the off-hook sequence begins:

  1. The off-hook state is detected by the digital switch.
  2. The switch establishes the time slot and sends a dial tone on the voice path.
  3. The switch awaits digits pressed by Party A.

The digit receipt sequence is as follows:

  1. Party A dials digits on the telephone handset.
  2. Each digit is received by the switch and sends a silence tone and starts Inter Digit Timer (IDT).
  3. IDT starts when the switch is awaiting a dialed digit and stops when the digit is pressed.

After Party A dials the last number, the ringdown sequence begins:

  1. When the digit receipt stops (or when the maximum dialed digits are pressed), the switch sends the request to the called number to allocate a time slot.
  2. When the called switch allocates a time slot the path is switched to the call handler.
  3. Party B’s phone rings (unless it is already off-hook, in which case a busy signal is routed back to party A).

Parties A and B can begin their conversation after the following sequence of steps is completed:

  1. Party B picks up the phone.
  2. The switch receives an answered call indication (off-hook).
  3. The ring-down signals stop.
  4. Parties A and B are able to speak on the established voice path.

After the two parties finish their conversation, the on-hook sequence of steps begins:

  1. The conversation ends with either party hanging up the phone.
  2. The on-hook indication is received by switches on access networks.
  3. The switches release established paths (termination).
  4. The call is ended.

The principles behind a VoIP call flow are similar to the call flow of POTS after it evolved into a digital system. The flows for VoIP involve sampling, quantizing and encoding. Instead of being transmitted over a circuit-switched network, the digital information is broken down into packets before transmission over a packet-switched network.

Now that we’ve discussed the POTS as an example of a circuit-switched network, we’ll briefly discuss in the next topic Circuit and Packet Switched Networks.


Circuit and Packet Switched Networks

Towards achieving the goal of efficiency in communications, it has been predicted that all forms of communication networks (Telephony, Television, Radio, Internet, etc.) will merge into one popular network that will carry all forms of communication. Merging all these forms of communication is called convergence.

Now, there is much evidence that packet-switched Internet Protocol (IP) is the layer over which current and future communication models, in fact almost everything else, will be carried.

Since the invention of the telephone by Antonio Meucci and Alexander Bell, circuit switching has been the most popular technology for voice communications.

Since the Advanced Research Projects Agency Network (ARPANET) adopted the Transmission Control Protocol / Internet Protocol (TCP/IP) in 1983 and the International Standards Organization (ISO) also created the Open Systems Interconnect (OSI) in 1984, packet switching has evolved substantially for digital data communications.

The two main switching techniques in use today are:

  • Circuit Switching used in Circuit-switched Networks and
  • Packet Switching used in Packet-switched Networks

Circuit Switching

The first switching technique used in communication networks was Circuit switching. And this is because it was simple enough to carry analog voice signals.

Even though the main example of its use is the Plain Old Telephone Systems (POTS), as earlier discussed, it is also used in the core of the internet in the form of Synchronous Digital Hierarchy (SDH) Synchronous Optical Networking (SONET) devices.

In circuit switching, the transmitting medium, typically copper in POTS, is mostly divided into channels using Time Division Multiplexing (TDM) or Frequency Division Multiplexing (FDM).

Circuit switching is inflexible because once a communication path is set, all parts of the transmission follow the same path.

All of the circuit’s bandwidth is reserved for the flow of information in circuit switching.

Towards ensuring that transmitted information is delivered in a timely fashion, the capacity of the circuit must at least be equal to the peak transmission rate of the flow. When this happens, the circuit is said to be peak-allocated, and then the network offers a connection-oriented service with an ideal Quality of Service (QoS).

Regardless of achieving this ideal QoS in POTS, bandwidth is simply wasted when sources idle or simply slow down. In addition, calls may be delayed or entirely blocked when there are not enough channels during the allocation of channels to circuits during call establishment.

In summary, circuit switching maintains a dedicated communication path or fixed bandwidth for transmission of information between sender and receiver even if there is no data being transmitted, thus inefficiently using bandwidth.


Packet Switching

Unlike the POTS, which uses circuit switching, VoIP uses packet switching technique. Before we discuss packet switching, the technology used in packet-switched networks, let’s first define what a packet is. A packet is the basic unit of data that is communicated between a sender and a receiver over a network. In order to achieve efficient transmission and minimize latency, the information to be transmitted in packet-switched networks is broken down into suitably-sized pieces called packets. Each transmitted packet contains a portion of user data and control information. These packets are sent, independently, to the nearest network node, which looks up the target address, and then forwards them to the next network node - a process that is repeated until the packets reach their destination.

Each packet makes its own way to the destination node and most of them may not even arrive in the same order in which they were sent from the sending station. At the receiving station, the packets are rearranged back into the original data that was sent.

Packet Switching uses a forwarding mechanism called Store and Forward technique during the transmission of packets. In this mechanism, packets are completely received, stored in the network node while being processed, and then transmitted to the next hop destination. If any of the network nodes between source and destination run out of buffer, packets are dropped.


Open Systems Interconnect (OSI) reference Model

A basic understanding of OSI and TCP/IP - how applications and nodes communicate over a network - is key in grasping potential sources of security issues and testing for vulnerabilities in VoIP.

The OSI model is a conceptual framework developed to serve as a guide for vendors and developers so that the products and systems they create can interoperate. This conceptual framework describes how applications communicate over a network.

The OSI model has seven logical layers, each one stacked upon the last.

Each of the seven layers handles a specific job and communicates with the layers below and above it.

Figure 2 – OSI Model


Seven Layers of the OSI Model

The seven abstraction layers of the OSI model can be defined as follows, from top to bottom. Our definition for all the layers shall assume the operation of a voice over internet protocol (VoIP) application such as linphone between users.

  • Layer 7 – Application layer

This is the layer at which the user interacts with the VoIP application – linphone. The user launches the application and then logs on with credentials.

  • Layer 6 – Presentation layer

This layer interacts with the application layer and also the layer below it. To improve speed and efficiency, it compresses data received from the application layer before delivering it to the layer below it – layer 5.  Another function for this layer is encryption. If the users communicating over linphone do that over an encrypted link, it is the responsibility of the presentation layer to add encryption from the sender’s side and then decode the encryption at the receiver’s side.

  • Layer 5 – Session layer

This layer is responsible for creating a communication when user A dials the extension number for the user B, and user B answers the call. The time between when this communication is opened and closed is known as a session. This layer also ensures that the session stays open long enough to transfer all voice and data being exchanged and promptly closes the session when either of the users sends a BYE signal. Another function of the session layer is synchronization of data transfer with checkpoints. Assume user A sends a 70MB file to user B through the linphone application. The session layer can set a checkpoint every 7MB so that if there is a link failure after 67MB of data is transferred, it can easily resume the session from the last checkpoint (63MB) without restarting the transmission of the whole file. This means only 7MB more of data needs to be transferred to the remote user.

  • Layer 4 –Transport layer

The function of the transport layer includes taking data from the session layer of the sender’s PC and breaking it into chunks called segments before handing it over to layer 3. On the receiving device, it does the reverse by reassembling the segments into data that the session layer can understand. Assuming the sender’s PC has a seventh-generation processor and transmits on a 100Mbps optical fibre link whereas the receiver’s PC has a Pentium 4 processor transmitting over a 1Mbps microwave link, the transport layer determines an optimal speed of transmission to ensure that the linphone application on the sender’s PC doesn't overwhelm the linphone application on the receiver's PC. This function of maintaining an optimal speed of transmission is called flow control.

  • Layer 3 – Network layer

In reality, the network layer is not required if the linphone application users are located within the same network (say, user A has the IP address 192.168.1.2/24 and user B has the IP address 192.168.1.3/24). In this case, the network layer is logically skipped and control is sent to the Data Link layer – layer 2. Otherwise, the network layer finds the best physical path for the voice call to move from user A to user B through a process known as routing. The network layer also breaks up segments from the transport layer into smaller units, called packets, on the sender's PC, and then reassembles these packets on the receiver’s PC.

  • Layer 2 – Data Link layer

The Data Link layer facilitates data transfer between two devices on the same network. It also takes packets from the network layer and breaks them into units called frames. The data link layer is made up of two sublayers – media access control (MAC) layer and logical link control (LLC) layer. The LLC provides multiplexing and flow control for the logical link whilst the MAC provides multiplexing and flow control for the transmission medium.

  • Layer 1 – Physical layer

This layer is responsible for the transmission and reception of bits from one device to the other. It also communicates with the transmission medium and defines the direction of transmission- simplex, half-duplex and full-duplex.

Figure 3 – OSI Model Layers and their corresponding protocols


Transmission Control Protocol / Internet Protocol (TCP/IP)

TCP/IP is a set of protocols or rules that govern the interconnection of network devices on the internet. In specifying how data is exchanged over the internet, it provides end-to-end communications that describe how data should be separated into packets before being addressed, transmitted, routed and received at the destination.

Figure 4 – How the TCP/IP model matches with the OSI model

The TCP/IP model divides its entire functions into four layers, where each layer performs a specific function. From top to bottom, they are defined as follows:

4. Application Layer:

The application layer of the TCP/IP protocol suite matches to the application layer of the OSI model. The presentation and session layers in the OSI model are not present in the TCP/IP model. Examples of protocols within the application layer of the TCP/IP domain are:

  • FTP/Telnet/SSH
  • HTTP/Secure HTTP (SHTTP)
  • POP3/SMTP
  • SNMP

3. Transport Layer

This is similar to the transport layer of the OSI model. The main entities used in the transport layer are TCP and User Datagram Protocol (UDP).

2. Internet Layer

The Internet layer of the TCP/IP model maps to the network layer of the OSI model. As such, the Internet layer is sometimes referred to as the network layer. It defines the entities that are responsible for logical transmission of data over the entire network, Internet Protocol - IP, Internet Control Message Protocol - ICMP and Address Resolution Protocol - ARP.

1. Link Layer

This is the lowest layer of the TCP/IP model. It is sometimes referenced as Network Access or Physical Access layer. And it maps to the Data link layer and Physical layer of the OSI model.


IP-Based Networks

The POTS system seems to be nearing the end of its lifecycle and operators are strategizing whether to replace this system for a potentially more viable path – migration towards IP-based networks.

It is now evident that systems and services can achieve their goals of operating more effectively and efficiently by migrating their processes to a model that uses IP-based standards that is independent from the underlying physical connection. The POTS network, for instance, first designed for transmitting analog voice signals from one customer to another, can now be used for accessing the internet at the same time a call is in session.

IP-based networks improve interoperability through a common occurrence known as convergence – a method which combines voice, video and data onto a single infrastructure.

In the commercial space, organizations have shifted to IP-based infrastructures in order to adapt to current trends.

The benefits of moving to IP-based networks include:

  1. The ability to run a variety of applications over a unified network
  2. Solve the problem of incompatibilities and
  3. Provide the foundation for convergence.

IP-based networks have provided the basis for the development of many applications including the Voice over IP (VoIP).

In the next module, we shall discuss the fundamentals of VoIP, its benefits, structure and relevant protocols.

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