Knowledge 

 
 

What is Network Monitoring?

Network Monitoring is a never-ending process that needs constant attention. In the IT jungle, one could easily get lost in the sea of devices and configuration settings. Our solution, is IT driven network management system, offers the best-in-class network monitoring with real-time monitoring and intelligent alerts that predict failures before they happen!  The only way to know if everything on a network is operating efficiently is with a network monitoring tool like GEMS.

 
 
 
 

Open Systems Interconnect (OSI) Model


Understanding of basic networking begins with the Open Systems Interconnect model.

The OSI model standardizes the key functions of a network using networking protocols. This allows different domains and devices types from different vendors to communicate with each other over a network. In the OSI model, network communications are grouped into seven logical layers.

The OSI Seven Layer Model

Why Monitor a Network?

Why is it important to monitor networks? Every company in the world has IT assets that enable faster, more efficient and more profitable business operations. IT assets create competitive advantages for companies but only if these systems is available. With many IT assets, there is no backup plan in the event of a loss. This is why the network is so important. It connects everything together and makes it possible for the IT infrastructure to function.

Networks are dynamic environments. GEMS is a solution for Network Admins to support new users, technologies and applications without impacting network performance. We've taken the guesswork out of managing your network with need-to-know information at your fingertips. It's the only network management solution that provides visibility to all areas of your network in one place.

When network problems arise, with GEMS is the new generation of root cause analysis. What is GEMS? GEMS is a proactive network monitoring tool that can quickly identify the root cause of network performance problems before they impact users, applications, and the business. You can use GEMS to pinpoint where in the network an issue is - so you can spend more time solving it and less time finding it.

 
 
 
 

How SNMP works

SNMP software agents communicate with network management systems to relay information on device and service status and configuration. The NMS provides a single interface for all your commands and alerts. You can batch multiple commands at once and get automatic notifications whenever something needs attention. SNMP overcomes shortcomings of other protocols thanks to the concept of a MIB.


The MIB is a technical description for describing the components and status information of a network device. MIBs can be created for any device in the IoT. They cover things like IP video cameras, vehicles, and industrial equipment. There's also an SNMP system that lets you monitor services such as Dynamic Host Configuration Protocol (DHCP) and ensure devices are running properly.


SNMP is a push and pull between network devices and the Network Management System (NMS). SNMP agents, who live with the MIB on a network device, collect information constantly but only push it over to the NMS when they are requested or when some aspect of the network crosses a pre-defined threshold known as a trap. Firing a trap is a great way to keep track of the issues that pop up. It allows you to remember those moments which could be important, errors or not.


SNMP also includes an inform message type that enables a network monitoring tool to acknowledge messages from a device. Trap is an SNMP operation that enables devices to send messages to SNMP management applications. The following is an example of a simple SNMP trap message:"management station HOST-1 sends trap message with text 'Hello world!' and hexadecimal value 0x0201." Inform messages allow agents to reset triggered alerts. Network management tools also can change a network device through the SNMP agent by sending a set message.


The NMS can also use a set message to make changes to a network device through the SNMP agent. This capability allows the network manager to make changes to the device configuration in response to new-network events. Often, SNMP functions in a synchronous model, in which the SNMP manager initiates the communication, and the agent responds to it. Typically, SNMP uses User Datagram Protocol (UDP) as its transport protocol. Well-known UDP ports for SNMP traffic are 161 (SNMP) and 162 (SNMPTRAP). They are the same in all versions of SNMP and should not change. SNMP commands can be used to perform tasks like resetting passwords, changing configuration settings, and reporting how much bandwidth, CPU and memory are in use.


SNMP is supported on a wide range of hardware. For example, you can find it in your router, switch, and wireless access point as well as in endpoints like your printer, scanner or IoT devices.

Components of SNMP

There are four main components to an SNMP-managed network. It includes the following:

  1. SNMP agent

Agent software is installed on the device or service that you want to monitor it. The software collects data about disk space, bandwidth use and other important network performance metrics. These agents use SNMP protocols to communicate with the manager. When queried by it, the agent sends the requested information back to the management system. If an error occurs, agents will proactively report it to the NMS. Most devices come with an SNMP agent pre-installed, but you usually need to set it up and turn it on.

  1. SNMP-managed network nodes

These are the network devices upon which agents run.

  1. SNMP manager

The NMS is a software platform that functions as a centralized console to which agents feed information about their environment and which the NMS gathers, aggregates, and displays data. The NMS may also be configured to send commands to agents. The NMS will actively request agents to send updates at regular periodically. The features available in the NMS will determine how widescale they can manage and monitor a network.

  1. Management information base

The MIB Database is a text file that lists all objects on a device that can be queried or controlled using SNMP. Every MIB item is given a unique OID.

SNMP commands

SNMP can perform a multitude of functions, usines a push and pull between network devices and the Network Management System (NMS). It can issue commands to read and write, such as resetting a password or changing a configuration setting. SNMP managers can monitor several indicators of performance such as CPU and memory metrics and automatically email or text alerts when specific thresholds are reached. Most of the time, SNMP operates in a synchronous model--it is initiated by the SNMP Manager and there is a response. These commands and messages, typically transported over TCP/IP or UDP, are known as protocol data units (PDUs).

These are some commonly used SNMP commands as follow.

  • GET Request: Sent by the SNMP manager and received by an agent, to obtain the value of a variable identified by its OID, in an MIB.
  • GETBULK Request: Sent by the SNMP manager to the agent to efficiently obtain a potentially large amount of data, especially large tables.
  • GETNEXT Request: Sent by the SNMP manager to the agent to retrieve the values of the next OID in the MIB's hierarchy.
  • INFORM Request: An asynchronous alert like a TRAP but requires confirmation of receipt by the SNMP manager.
  • RESPONSE: Sent by the agent to the SNMP manager, issued in reply to a GET Request, GETNEXT Request, GETBULK Request and a SET Request. Contains the values of the requested variables.
  • SET Request: Sent by the SNMP manager to the agent to issue configurations or commands.
  • TRAP: An asynchronous alert sent by the agent to the SNMP manager to indicate that something significant event has happened, such as an error or failure, has occurred.

What are SLAs and why are they important?


The SLA is a contract between IT and Line of Business Owners that commits to providing a certain level of network performance and uptime. It is often created after a business-wide assessment, so it can be tailored to the needs of each individual organization. SLAs are a commitment by a service provider to their customer that the provider will deliver a predefined level of service. SLAs are often used to provide a level of performance guarantee. They are normally measured and reported on for performance management purposes, and might be tied to IT compensation plans.

Why are SLAs important? Because poor performance and down time is costly. For an e-Commerce website like Amazon, one hour of downtime can cost millions in lost revenue.

 
 
 

What are SLAs and why are they important?


Companies across the world use a network monitoring system to ensure security, prevent downtime and protect against cyber-attacks. Network monitoring systems poll network devices and servers for performance data using standard protocols such as SNMP, WMI, NetFlow and more. These systems are an integral part of any business's IT infrastructure using standard protocols such as:

  • SNMP, Simple Network Management Protocol
  • WMI, Windows Management Instrumentation
  • SSH, Secure Shell for Unix and Linux server
  • The two most widely used monitoring protocols are SNMP and WMI.
  • Simple Network Management Protocol (SNMP)

SNMP is an Internet Standard protocol for collecting and organizing information about managed devices on IP networks and for modifying that information to change device behaviour. Devices that typically support SNMP include cable modems, routers, switches, servers, workstations, printers, and moreSNMP is standard protocol that collects data from almost any network attached device, including: Routers, Switches, Wireless LAN Controllers, wireless Access Points, Servers, Printers and more.

SNMP works by querying “Objects”. An object is something that an NMS collects information about. For instance, CPU utilization is an SNMP object. Querying on the CPU utilization object would return a value that an NMS uses for alerting and reporting.

The Objects queried by SNMP are maintained in a Management Information Base, or MIB. The Management Information Base (MIB) can be thought of as a database of managed objects that the agent tracks. Any sort of status or statistical information that can be accessed by the NMS is defined in a MIB.  For example, the MIB for a Cisco router will contain all objects, defined by Cisco, that can be used to monitor that router such as CPU utilization, memory utilization and interface status.The objects in a MIB are catalogued using an standardized numerating system. Each object has its own, unique Object Identifier, or OID.

Some NMSs provides a MIB Browser. A MIB Browser allows Network Admins navigate thru a MIB to find additional objects that they want to monitor on a device.

Windows Management Instrumentation (WMI)

WMI (Windows Management Instrumentation); It is a technology that enables almost every object to be controlled in Windows operating systems and can perform operations and management functions in the operating system. This protocol creates an operating system interface that receives information from devices running a WMI agent. The WMI gathers details about the operating system, hardware or software data, the status and properties of systems, configuration and security information, and process and services information. Then sent all of these detail data along to the network management software, which monitors network health, performance, and availability.

 
 
 
 
 

What is VPN

VPN stands for Virtual Private Network. It is a method by which two end-points create a single, private connection, or tunnel, while using a larger network infrastructure such as the internet or wide area network. When established, a VPN acts like a direct connection to a private network. The VPN itself simply acts like a network interface to the client and is transparent the operating system, applications, and users accessing the VPN network. Therefore, applications, messages, and users can all use the connection normally without any need to understand how the VPN operates.

How to create a VPN

 A traditional VPN requires two endpoints. One is the remote endpoint and the other is the local endpoint. To create the VPN connection, on both endpoints must match or support the VPN methodology used on the other endpoint. Once both endpoints have been established VPN connection, they create a connection called a VPN tunnel.

Types of VPN

Remote client

In this type of VPN, the remote user must have a VPN client installed and configured to connect to a VPN gateway on the local network.

Site-to-site

Another commonly used form of VPN allows for a WAN-style connection between two different sites by using a public network such as the internet rather than going through the expense and difficulty of installing a direct, private connection.

Client-to-provider

A VPN, which connects you to a VPN provider who in turn has a network connection to the internet, is becoming increasingly common. The user must have a VPN client installed and configured to connect to the remote VPN provider’s VPN servers. The VPN connection only lasts for the first part of the connection and not all the way to the destination.

VPN components

 A VPN works by establishing a secure, point-to-point connection between the remote client and a VPN server connected to the target network. A VPN connection establishes a secure channel to route data & IP header on the local network. Once established, it encapsulates and encrypts both behind the remote endpoint.. An IP header designed to route across the insecure, public network is added, and then the data is ready for send it across a public network.

VPN clients

Standalone VPN clients

Standalone VPN clients require the installation of software on one or both endpoints. In-order to establish a VPN connection, the endpoint must run the VPN client and connect to the other endpoint.

Built-in OS VPN Clients

All major operating systems, including Windows, iOS, MacOS, Android and Linux make it easy to connect to third-party VPN servers. It's just a matter of ensuring the remote VPN endpoint supports the same protocol and configurations. These clients are often not easily configured by non-technical users. They are therefore most often used in a corporate environment.

VPN server

A remote endpoint that connects to a VPN server also typically features a client which supports all the VPN server settings. Typically, this means that the VPN sever should act as both a gateway and router (at least) at the edge of one's local network or computer system. The server is responsible for unfolding the packets and reshaping them so they can be distributed to the local network or internet.

Tunneling protocol

A VPN which aims to connect you to a public network must establish a regular non-VPN connection with that network. This is done by way of tunneling protocols. A tunneling protocol (e.g., the Virtual Private Network Protocol) wraps each transmitted packet such that it can be read and transmitted across networks that are not private.

                                                                What is an IP address?

An IP address is a unique identifier for every machine using the Internet. Known as your “internet protocol address,” this identifier is written as a string of numbers separated by periods. The Internet Protocol (IP) is part of the Internet layer of the Internet protocol suite. In the OSI model, IP would be considered part of the network layer. IP is traditionally used in conjunction with a higher-level protocol, most notably TCP. The IP standard is governed by RFC 791.

Most addresses are IPv4. It’s the most widely-deployed IP used to connected devices to the Internet. Addresses in IPv4 are 32-bits long. This allows for a maximum of 4,294,967,296 (232) unique addresses. IPv6 is the most recent version Addresses in IPv6 are 128-bits, which allows for 3.4 x 1038 (2128) unique addresses. IP addresses are binary numbers but are typically expressed in decimal form (IPv4) or hexadecimal form (IPv6) to make reading and using them easier for humans.

 
 
 
 

                                                                  How IP works

An IP address is a unique identifier assigned to a device or domain that connects to the Internet. This means that IP must work without a central directory or monitor, and that it cannot rely upon specific links or nodes existing. IP is a connectionless protocol that is datagram-oriented., so each data packets must contain the source IP address, destination IP address, and other data in the header to be successfully delivered. Combined, these factors make IP an unreliable, best effort delivery protocol. An IP error correction is handled by upper level protocols instead. These protocols include TCP, which is a connection-oriented protocol, and UDP, which is a connectionless protocol. Most Internet traffic is TCP/IP.

 
 
 
 
 

IP versions

IPv4 address

The original IPv4 specification was designed for the ARPANET in 1983 network that would eventually become the internet. Originally a test network, no one contemplated how many addresses might be needed in the future. At the time, the 232 addresses (4.3 billion) were surely considered sufficient. However, over time, it became obvious that as currently implemented, the IPv4 address space would not be big enough for a worldwide internet with numerous connected devices per person.

IPv6 addresses

o avoid the seemingly reoccurring issue in technology, where a specification’s limitation seems more than sufficient at the time, but inevitably becomes too small, the designers of IPv6 created an enormous address space for IPv6. The address size was increased from 32 bits in IPv4 to 128 bits in IPv6.

The IPv6 has a theoretical limit of 3.4 x 1038 addresses. That’s over 340 undecillion (340 trillion trillion trillion) addresses, which is make sure that every single atom on earth has one IP address.

What is DNS server

On the Internet, name resolution is handled by the Domain Name System (DNS). DNS is a directory of names that match with numbers. The numbers, in this case are IP addresses, which computers use to communicate with each other. When the connection is initiated, the source host will request the IP address of the destination host from a DNS server. The DNS server will reply with the destination’s IP address. This IP address will then be used for all communications sent to that name.

                                                                  What is Ping?

Ping  (Packet Internet or Inter-Network Groper) is a computer network administration utility command to test available on virtually any operating system with network connectivity, that acts as a test to see if a networked device is reachable.

The ping command sends a request over the network to a specific device. A successful ping results in a response from the computer that was pinged back to the originating computer.

Ping and speed test

The term is also used to test and determine how fast a data signal travels from one place, like a computer, to another, like a website. 

 
 
 

                                                                 How to use Ping

Ping works by sending an Internet Control Message Protocol (ICMP). The ping a command-line based utility, ping lends itself to easy use in various scripts, allowing for numerous pings to run and be recorded for all type of usage.

Troubleshooting with Ping

Perhaps the most common use of the ping utility is in troubleshooting. When trying to use applications or systems over a network, the most important thing to know is if there is actually a working connection or device. When a ping command is issued, a ping signal is sent to a specified address. When the target host receives the echo request, it responds by sending an echo reply packet. A quick ping by IP address will confirm that the system is on, that there is a connection, and that the two machines can talk to each other.

Ping IP address

Ping error

If ping gets no response from the target host, most implementations of ping display nothing or a timeout notification. The result may look like this, for example:

  • Pinging 121.242.124.9 with 32 bytes of data:
  • Request timed out.
  • Request timed out.
  • Request timed out.
  • Request timed out.

Discovery

Ping can be used as a quick and dirty discovery tool. Since virtually any network connected device will respond to a ping, pinging a range of addresses.

Monitoring

Ping can be used to monitor the network availability of devices. When a ping command ıssued as a scheduled task can offer simple polling of any networked computer or device without the need to install any additional software agents, and without the need to open additional ports. The simplest any up/down monitor could be accomplished by running a ping until stopped. When the pings start failing, there is an issue reaching the link or device

Measuring Latency

round-trip time (RTT) or latency, is a measure of how long it took to receive a response. Measured in milliseconds (ms), the process starts when a when pink request issued to a server and is completed when a response from the server is received packet.

 
 
 

                                                                 What is bandwidth?

Bandwidth (BW) is measured as the amount of data that can be transmitted from one point to another within a network in a specific period. And bandwidth is expressed as a bitrate and measured in bits per second (bps).

The term bandwidth in general implies capacity. Bandwidth is also a key concept in several other technological fields. In a transmission such as a radio signal and is typically measured in hertz (Hz).

Expressing bandwidth

As mentioned earlier, Bandwidth is measured in bits per second and expressed in bps. However, today's networks typically have capacities that are too large to be expressed in bps, so they cannot be easily expressed using such small units. It's now common to see higher numbers denoted by the metric power of 10, such as Mbps (megabits per second), Gbps (gigabits per second), or Tbps (terabits per second).

  • K = kilo = 1,000 bits. =10³
  • M = mega = 1,000 kilo = 1,000,000 bits.=10⁶
  • G = giga = 1,000 mega = 1,000,000,000 bits. =10
  • T = tera = 1,000 giga = 1,000,000,000,000 bits.=10¹²

After terabit, there are petabit, exabit, zettabit, and yottabit, each representing an additional power of 10.

                                                                       What is NetFlow?

NetFlow is a network protocol developed by Cisco Systems that allows administrators to collect and analyze data on network traffic flows. It is often used for network performance monitoring and troubleshooting, security, and capacity planning.

 

NetFlow works by capturing information about each individual flow of traffic passing through a network device, such as a router or switch. This information includes the source and destination IP addresses, port numbers, and protocol. The collected data is stored in a flow record and can be analyzed to gain insights into network traffic patterns and trends.

 

There are several versions of the NetFlow protocol, including NetFlow v1, v5, v7, v8, and v9. Each version has its own set of features and capabilities and is designed to meet the specific needs of different environments and use cases.

 

NetFlow data is typically collected by a NetFlow collector, which is a separate server or computer running a NetFlow receiver. NetFlow data is typically exported using UDP and sent to the collector's IP address and a configured destination port. In some cases, SNMP can be used to turn on NetFlow and configure the collector's IP address.

 

NetFlow data can also be accessed through the NetFlow MIB, which allows for the gathering of data via SNMP. This data includes upstream/downstream traffic and packets/bytes per flow, as well as information on "top talkers" and other traffic types.

 

                                              What Is a Network Diagram?

A network diagram provides a visual representation of a network, displaying how the individual elements work interact. . Depending on the scope and purpose, a network diagram may provide a simple overview of the network or a more detailed network diagram. There are two main types of network diagrams: physical and logical. Physical network diagrams.

Physical Network Diagrams

A physical network diagram, as its name suggests, shows the actual physical arrangement of the network elements and cable connections. Physical network diagrams shows all of the physical aspects of the network,  likely include: ports, cables, racks, servers, specific models, and so on. Within the OSI model of networking, physical diagrams are referred to as ‘L1’

Logical Network Diagrams

A logical network diagram (Layer 3) addresses how information in the network flows. This means that you’ll generally visualize the following elements in your logical network topology:

  • subnets (such as: IP addresses, VLAN IDs, and subnet masks,)
  • network objects (routers and firewalls)
  • network segments
  • specific routing protocols
  • routing domains
  • traffic flow
  • voice gateways
 

 

The Erlang B formula was developed by the Danish mathematician Agner Krarup Erlang. He was a pioneer in the field of telecommunication engineering, and his work laid the foundations for many of the techniques and principles that are used in modern communication systems. In addition to the Erlang B formula, he also developed the

Erlang C formula, which is used to calculate the required number of channels in a system to achieve a desired level of service quality.

Overall, Erlang's contributions to the field of telecommunication engineering have had a lasting impact, and his work is still widely used and referenced in the telecom industry today.

The Erlang B theory is a mathematical model used to predict the performance of a telephone switching system or another communication system. It is based on the assumption that calls arrive at a switch or other communication system according to a Poisson process, which is a type of statistical process characterized by a constant arrival rate and a random distribution of interarrival times.

In the telecom industry, the Erlang B theory is used to design and dimension communication systems, as well as to predict their performance and capacity. It is often used in conjunction with the grade of service (GoS), which is a measure of the probability that a call will be blocked due to insufficient resources. The GoS can be calculated using the Erlang B formula, which allows engineers to determine the required number of channels (i.e., available resources) in a system to achieve a desired GoS.

The Erlang B formula is given by:

B = (A^N)/(N! * (N-A)!)

where:

B is the probability of blocking,

A is the traffic intensity in erlangs,

N is the number of channels.

For example, if a system has a desired GoS of 0.2% and a traffic intensity of 10 erlangs, we can use the Erlang B formula to calculate the required number of channels as follows:

B = 0.002 A = 10 erlangs N = ?

Substituting the values into the formula, we get: 0.002 = (10^N)/(N! * (N-10)!)

Solving this equation for N, we find that the required number of channels is approximately 16.7.

Overall, the Erlang B theory is an important tool in the telecom industry for predicting and managing the performance and capacity of communication systems. It is used to ensure that systems are designed to handle the expected levels of traffic and achieve the desired GoS, and identify potential bottlenecks or other issues that may impact their performance.

 

  • Erlangs: In telecommunications, traffic intensity is often measured in units called Erlangs. An Erlang is a unit of traffic intensity that is defined as the product of the average call duration and the average number of calls per unit of time. For example, if a network receives 100 calls per hour, each lasting an average of 10 minutes, the traffic intensity would be 10 Erlangs (100 calls x 10 minutes/call).

 

  • Bits per second: In some contexts, traffic intensity may be measured in terms of the amount of data being transmitted over a network, typically expressed in bits per second (bps). For example, if a network is transmitting 100 megabits per second (Mbps) of data, its traffic intensity would be 100 Mbps.

 

  • Calls per second: In some cases, traffic intensity may be measured in terms of the number of calls being made over a network per unit of time, typically expressed in calls per second (cps). For example, if a network is receiving 10 calls per second, its traffic intensity would be 10 cps.

 

  • Mean holding time: In some contexts, traffic intensity may be defined as the mean holding time of calls or communication sessions on a network, typically expressed in seconds. For example, if the average call duration on a network is 10 minutes, the traffic intensity would be 600 seconds (10 minutes x 60 seconds/minute).

Overall, traffic intensity is a measure of the amount of communication activity taking place on a network, and it can be defined in several different ways depending on the specific context in which it is being used. Mean holding time is one way in which traffic intensity can be defined, and it can be useful for understanding the amount of time that users are spending on calls or other communication sessions on a network.

 

Traffic intensity and traffic carried are two related but distinct concepts in telecommunication engineering.

Traffic intensity is a measure of the average number of calls or other communication events per channel per unit of time in a communication system. It is typically expressed in units of erlangs, which is defined as the product of the traffic intensity and the total time period over which the traffic is measured.

For example, if a switch handles an average of 10 calls per hour and operates for 8 hours per day, the traffic intensity in erlangs would be 10 * 8 = 80 erlangs.

Traffic carried, on the other hand, is a measure of the total amount of traffic (i.e., calls or other communication events) that is handled by a communication system over a given time period. It is typically expressed in units of erlang-hours, which is defined as the product of the traffic intensity and the time period over which the traffic is measured.

For example, if a switch handles an average of 10 calls per hour and operates for 8 hours per day, the traffic carried in erlang-hours would be 10 * 8 = 80 erlang-hours.

Overall, traffic intensity is a measure of the average load on a communication system, while traffic carried is a measure of the total amount of traffic that the system handles over a given time period. Both concepts are important in the design and dimensioning of communication systems, as they help to determine the required capacity and performance of the system.

In telecommunications, the term "traffic carried" refers to the total volume of data or communication sessions that are transmitted over a network during a given period of time. It is a measure of the amount of communication activity taking place on the network, and it can be used to assess the capacity and performance of the network.

"Mean traffic," on the other hand, refers to the average volume of traffic being transmitted over a network during a given period of time. It is calculated by dividing the total traffic carried by the number of time units over which it was transmitted. For example, if a network carried 1,000 megabits of traffic over a period of 10 hours, its mean traffic would be 100 megabits per hour (1,000 megabits / 10 hours).

Mean traffic is often used as a measure of the average load on a network or system, and it can be useful for understanding the typical traffic patterns on a network over time. It is often used in conjunction with other measures of traffic, such as busy hour traffic and peak traffic, to provide a more complete picture of the traffic patterns on a network.

The Erlang B theory is a mathematical model used to predict the performance of a telephone switching system or other communication system. It is based on the assumption that calls arrive at a switch or other communication system according to a Poisson process, which is a type of statistical process characterized by a constant arrival rate and a random distribution of interarrival times.

The Erlang B theory is used to calculate the probability that a call will be blocked (i.e., unable to be completed due to insufficient resources) as a function of the number of channels (i.e., available resources) in the system, the traffic intensity (i.e., the average number of calls per channel per unit of time), and other factors. This probability is known as the grade of service (GoS), which is a measure of the likelihood that a call will be blocked.

In a telephone switching system, congestion occurs when the number of calls arriving at the system exceeds the number of available channels, resulting in a high GoS and a higher probability of call blocking. When congestion occurs, calls may be queued or dropped, leading to a degradation in the quality of service (QoS) experienced by users.

To avoid congestion and ensure a high QoS, telecom companies must carefully design and dimension their communication systems to ensure that they have sufficient capacity to handle the expected levels of traffic. This can be done using tools such as the Erlang B theory and other traffic engineering techniques.

Overall, the Erlang B theory is an important tool in the telecom industry for predicting and managing the performance and capacity of communication systems, and for minimizing the likelihood of congestion and call blocking. It is used to ensure that systems are designed to handle the expected levels of traffic and to achieve the desired GoS, and to identify potential bottlenecks or other issues that may impact their performance.

The relationship between quality of service (QoS) and system capacity is complex, and it can vary depending on the specific characteristics of a given network or system. In general, however, there is a strong correlation between QoS and system capacity, as the capacity of a system can have a significant impact on its ability to deliver high-quality communication services to users.

As a general rule, increasing the capacity of a system will typically result in an improvement in QoS. For example, if a telecommunications network has a capacity of 100 megabits per second (Mbps) and is transmitting 50 Mbps of traffic, it will generally be able to deliver higher quality communication services to users than if it had a capacity of 50 Mbps and was transmitting 50 Mbps of traffic.

However, it is important to note that simply increasing capacity is not always sufficient to improve QoS, as there are other factors that can also impact the quality of communication services. For example, network congestion, interference, and other external factors can all affect QoS, even if the system has sufficient capacity to handle the traffic.

Overall, there is a strong correlation between QoS and system capacity, and increasing the capacity of a system can generally lead to an improvement in QoS. However, other factors such as network congestion and interference can also impact QoS, and it is important to consider these factors when seeking to optimize the quality of communication services.

The GoS of a network can be calculated using the following formula:

GoS = (Carried traffic / Offered traffic) x 100%

This formula expresses GoS as a percentage, with a higher percentage indicating a better GoS. For example, a GoS of 90% would mean that 90% of the traffic that is attempted to be transmitted over the network is successfully carried, while 10% is lost.

GoS is an important factor to consider in the operation and optimization of a communication network, as it directly affects the user experience and the quality of the services provided. Network administrators can use this formula to monitor GoS and identify strategies for improving it, such as by optimizing the allocation of resources or implementing traffic management techniques.

  • Call set-up time: The call set-up time is the amount of time that it takes to establish a connection between two parties for a communication session or call. It includes the time required to initiate the call, negotiate the communication parameters, and establish the connection.
  • Conversation time: The conversation time is the amount of time that the two parties spend talking or communicating during a call. It is the duration of the call from the point at which the connection is established until one of the parties disconnects.
  • Disconnection time: The disconnection time is the amount of time that it takes to terminate a call or communication session. It includes the time required to release the connection and any other resources used during the call.
  • Occupation time: The occupation time is the total amount of time that a communication channel or resource is occupied by a call or communication session. It includes the call set-up time, conversation time, and disconnection time.

As for mean holding time, it is a measure of the average duration of calls or communication sessions on a network. It is calculated by dividing the total holding time of all calls or sessions by the number of calls or sessions. For example, if a network has 10 calls that last a total of 100 minutes, its mean holding time would be 10 minutes (100 minutes / 10 calls). Mean holding time is often used as a measure of traffic intensity in telecommunications, as it helps to understand the amount of time that users are spending on calls or other communication sessions on a network.

  • In telecommunications, the term "loss system" refers to a system or network that is designed to handle a certain volume of traffic, but is unable to deliver all of the traffic that is offered to it. This can occur when the volume of traffic exceeds the capacity of the system, resulting in some of the traffic being lost or dropped.

    Loss systems can be characterized by the probability of loss, which is a measure of the likelihood that a given call or communication session will be lost or dropped due to capacity constraints. The probability of loss is typically expressed as a percentage, and it is calculated by dividing the number of calls or sessions that are lost by the total number of calls or sessions offered to the system.

    Loss systems can be contrasted with blocking systems, which are systems that are designed to reject or block any additional traffic once the system reaches capacity. In a blocking system, all calls or communication sessions that are offered to the system beyond its capacity will be blocked or rejected, rather than being lost or dropped.

    Overall, loss systems are a common occurrence in telecommunications networks, and they can impact the quality of communication services for users. To mitigate the effects of loss systems, telecommunications providers may use techniques such as traffic engineering and capacity planning to optimize the use of network resources and minimize the probability of loss.

In telecommunications, a delay system is a system or network that is designed to handle a certain volume of traffic, but is unable to deliver it in real time. This can occur when the volume of traffic exceeds the capacity of the system, resulting in delays in the transmission of the traffic.

Delay systems can be characterized by the delay experienced by users, which is a measure of the amount of time that it takes for a given call or communication session to be transmitted over the system. Delay can be caused by a variety of factors, including network congestion, interference, and other capacity constraints.

Delay systems can be contrasted with loss systems, which are systems that are designed to handle a certain volume of traffic but are unable to deliver all of the traffic that is offered to them. In a loss system, some of the traffic is lost or dropped due to capacity constraints, while in a delay system, all of the traffic is transmitted but with a delay.

Overall, delay systems are a common occurrence in telecommunications networks, and they can impact the quality of communication services for users. To mitigate the effects of delay systems, telecommunications providers may use techniques such as traffic engineering and capacity planning to optimize the use of network resources and minimize the delay experienced by users.

Loss systems and delay systems can occur in various types of telecommunications networks and systems, including telephone networks, mobile networks, data networks, and other types of communication systems. They can occur in both wired and wireless networks, and they can be caused by a variety of factors, including network congestion, interference, and other capacity constraints.

Here are a few examples of where loss systems and delay systems may occur today:

  • Telephone networks: Loss systems and delay systems can occur in traditional telephone networks, such as those used for landline and VoIP (Voice over Internet Protocol) calls. They can be caused by factors such as network congestion and capacity constraints, and they can impact the quality of communication services for users.
  • Mobile networks: Loss systems and delay systems can also occur in mobile networks, such as those used for cellular phone calls and data transmission. They can be caused by factors such as network congestion, interference, and limited capacity, and they can impact the quality of communication services for mobile users.
  • Data networks: Loss systems and delay systems can occur in data networks, such as those used for internet connectivity and data transmission. They can be caused by factors such as network congestion, limited capacity, and interference, and they can impact the performance and reliability of data transmission.

Overall, loss systems and delay systems can occur in a variety of telecommunications networks and systems, and they can impact the quality and performance of communication services for users. Telecommunications providers use various techniques and strategies to minimize the occurrence of loss systems and delay systems, and to ensure that networks are able to deliver high-quality communication services to users.

 

Yes, subscriber behaviour is an important factor to consider in telecommunications, as it can have a significant impact on the performance and capacity of a network. The way in which subscribers react to unsuccessful call attempts, such as by making a new attempt, can affect the volume of traffic on the network and the probability of loss or delay.

In general, if subscribers react to unsuccessful call attempts by making a new attempt, it can increase the volume of traffic on the network and potentially lead to more instances of loss or delay. This is because each new call attempt consumes additional network resources, and if the volume of traffic exceeds the capacity of the network, it can result in lost or delayed calls.

To mitigate the impact of subscriber behaviour on the performance and capacity of a network, telecommunications providers may use various techniques, such as traffic engineering and capacity planning, to optimize the use of network resources and ensure that the network is able to handle the expected volume of traffic effectively. They may also implement strategies such as network congestion control algorithms to manage the flow of traffic and minimize the probability of loss or delay.

Erlang's first formula, also known as Erlang's loss formula, can be used to calculate the probability of congestion in a loss system, given the volume of traffic offered to the system in Erlangs. The formula is as follows:

Probability of congestion = (Traffic offered in Erlangs) / (Capacity in Erlangs)

For example, if a network has a capacity of 10 Erlangs and is being offered a traffic volume of 15 Erlangs, the probability of congestion would be calculated as follows:

Probability of congestion = (15 Erlangs) / (10 Erlangs) = 1.5

This indicates that there is a 1.5 times greater probability of congestion occurring on the network due to the volume of traffic being offered exceeding the capacity of the network.

Erlang's second formula, also known as Erlang's delay formula, can be used to calculate the probability of delay in a delay system, given the volume of traffic offered to the system in Erlangs. The formula is as follows:

Probability of delay = (Traffic offered in Erlangs)^2 / (2 Capacity in Erlangs (Capacity in Erlangs - Traffic offered in Erlangs))

For example, if a network has a capacity of 10 Erlangs and is being offered a traffic volume of 15 Erlangs, the probability of delay would be calculated as follows:

Probability of delay = (15 Erlangs)^2 / (2 10 Erlangs (10 Erlangs - 15 Erlangs)) = 225 / (2 10 (-5)) = 225 / (-100) = -2.25

This indicates that there is a 2.25 times greater probability of delay occurring on the network due to the volume of traffic being offered exceeding the capacity of the network.

It's important to note that both Erlang's first and second formulas are based on assumptions about the behaviour of traffic in loss and delay systems, and they may not always provide accurate results in all cases. In practice, it may be necessary to use more advanced tools and techniques to accurately model and analyse the performance of loss and delay systems.

Erlang's first formula, also known as Erlang's loss formula, can be used to calculate the probability of congestion in a loss system, given the volume of traffic offered to the system in Erlangs. The formula is as follows:

Probability of congestion = (Traffic offered in Erlangs) / (Capacity in Erlangs)

For example, if a network has a capacity of 10 Erlangs and is being offered a traffic volume of 15 Erlangs, the probability of congestion would be calculated as follows:

Probability of congestion = (15 Erlangs) / (10 Erlangs) = 1.5

This indicates that there is a 1.5 times greater probability of congestion occurring on the network due to the volume of traffic being offered exceeding the capacity of the network.

Erlang's second formula, also known as Erlang's delay formula, can be used to calculate the probability of delay in a delay system, given the volume of traffic offered to the system in Erlangs. The formula is as follows:

Probability of delay = (Traffic offered in Erlangs)^2 / (2 Capacity in Erlangs (Capacity in Erlangs - Traffic offered in Erlangs))

For example, if a network has a capacity of 10 Erlangs and is being offered a traffic volume of 15 Erlangs, the probability of delay would be calculated as follows:

Probability of delay = (15 Erlangs)^2 / (2 10 Erlangs (10 Erlangs - 15 Erlangs)) = 225 / (2 10 (-5)) = 225 / (-100) = -2.25

This indicates that there is a 2.25 times greater probability of delay occurring on the network due to the volume of traffic being offered exceeding the capacity of the network.

It's important to note that both Erlang's first and second formulas are based on assumptions about the behaviour of traffic in loss and delay systems, and they may not always provide accurate results in all cases. In practice, it may be necessary to use more advanced tools and techniques to accurately model and analyse the performance of loss and delay systems.

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