We will be covering most of the certification.
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We are starting with Cvoice exam, after passing it you get CCNA voice title.
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We will be covering most of the certification.
Best of luck for your exams and lets get set go.
We are starting with Cvoice exam, after passing it you get CCNA voice title.
CCENT
Exam Description
CCENT 640-822 ICND1
Associated Certifications: CCENT and CCNA
Duration: 90 minutes (40-50 questions)
The 640-822 Interconnecting Cisco Networking Devices Part 1 (ICND1) is the exam associated with the Cisco Certified Entry Network Technician certification and a tangible first step in achieving the Cisco Certified Network Associate certification. This exam tests a candidate's knowledge and skills required to successfully install, operate, and troubleshoot a small branch office network. The exam includes topics on networking fundamentals; connecting to a WAN; basic security and wireless concepts; routing and switching fundamentals; the TCP/IP and OSI models; IP addressing; WAN technologies; operating and configuring IOS devices; configuring RIPv2, static and default routing; implementing NAT and DHCP; and configuring simple networks.
Exam Topics
Describe the operation of data networks
Describe the purpose and functions of various network devices 1 2 3
Select the components required to meet a given network specification 1 2 3
Use the OSI and TCP/IP models and their associated protocols to explain how data flows in a network 1 2
Describe common networking applications including web applications
Describe the purpose and basic operation of the protocols in the OSI and TCP models 1 2
Describe the impact of applications (Voice Over IP and Video Over IP) on a network 1 2 3
Interpret network diagrams
Determine the path between two hosts across a network 1 2 3
Describe the components required for network and Internet communications
Identify and correct common network problems at layers 1, 2, 3 and 7 using a layered model approach
Differentiate between LAN/WAN operation and features
Implement a small switched network
Select the appropriate media, cables, ports, and connectors to connect switches to other network devices and hosts
Explain the technology and media access control method for Ethernet technologies
Explain network segmentation and basic traffic management concepts
Explain the operation of Cisco switches and basic switching concepts
Perform, save and verify initial switch configuration tasks including remote access management
Verify network status and switch operation using basic utilities (including: ping, traceroute,telnet,SSH,arp, ipconfig), SHOW & DEBUG commands
Implement and verify basic security for a switch (port security, deactivate ports)
Identify, prescribe, and resolve common switched network media issues, configuration issues, autonegotiation, and switch hardware failures
Implement an IP addressing scheme and IP services to meet network requirements for a small branch office
Describe the need and role of addressing in a network " Create and apply an addressing scheme to a network
Assign and verify valid IP addresses to hosts, servers, and networking devices in a LAN environment
Explain the basic uses and operation of NAT in a small network connecting to one ISP
Describe and verify DNS operation
Describe the operation and benefits of using private and public IP addressing
Enable NAT for a small network with a single ISP and connection using SDM and verify operation using CLI and ping
Configure, verify and troubleshoot DHCP and DNS operation on a router.(including: CLI/SDM)
Implement static and dynamic addressing services for hosts in a LAN environment
Identify and correct IP addressing issues
Implement a small routed network
Describe basic routing concepts (including: packet forwarding, router lookup process)
Describe the operation of Cisco routers (including: router bootup process, POST, router components)
Select the appropriate media, cables, ports, and connectors to connect routers to other network devices and hosts
Configure, verify, and troubleshoot RIPv2
Access and utilize the router CLI to set basic parameters
Connect, configure, and verify operation status of a device interface
Verify device configuration and network connectivity using ping, traceroute, telnet, SSH or other utilities
Perform and verify routing configuration tasks for a static or default route given specific routing requirements
Manage IOS configuration files (including: save, edit, upgrade, restore)
Manage Cisco IOS
Implement password and physical security
Verify network status and router operation using basic utilities (including: ping, traceroute,telnet,SSH,arp, ipconfig), SHOW & DEBUG commands
Explain and select the appropriate administrative tasks required for a WLAN
Describe standards associated with wireless media (including: IEEE WI-FI Alliance, ITU/FCC)
Identify and describe the purpose of the components in a small wireless network. (including: SSID, BSS, ESS)
Identify the basic parameters to configure on a wireless network to ensure that devices connect to the correct access point
Compare and contrast wireless security features and capabilities of WPA security (including: open, WEP, WPA-1/2)
Identify common issues with implementing wireless networks
Identify security threats to a network and describe general methods to mitigate those threats
Explain today's increasing network security threats and the need to implement a comprehensive security policy to mitigate the threats
Explain general methods to mitigate common security threats to network devices, hosts, and applications
Describe the functions of common security appliances and applications
Describe security recommended practices including initial steps to secure network devices
Implement and verify WAN links
Describe different methods for connecting to a WAN
Configure and verify a basic WAN serial connection
Recommended Training
The following course is the recommended training for this exam:
Interconnecting Cisco Networking Devices Part 1 (ICND1) v1.0
The course listed is offered by Cisco Learning Partners the authorized source for Cisco IT training delivered exclusively by Certified Cisco Instructors. For a list of Cisco Learning Partners, use the Learning Partner Locator.
Additional Resources
A variety of Cisco Press titles may be available for this exam. These titles can be purchased through the Cisco Marketplace Bookstore,or directly from Cisco Press.
Describe the purpose and functions of various network devices 1
What is a Router?
Routers operate at Layer3 of the OSI Layer and are categorized in the Network Layer. A router can be programmed to find the best routes between networks. The router can traverse many networks and is capable of choosing the best route to the destination. Offices in different and remote locations can be connected together using routers using dedicated or switched lines. The lines are usually provided by telephone companies or Internet Service Providers (ISPs). The routers connect to lines using serial interfaces which usually come installed in the device. Connecting the lines to the serial interfaces on the router allow for a Wider Area Network (WAN) connection.
For example, if you have one office in New York and other in London then connecting in the above way will allow for the sending of data and voice transmissions. The Internet is made up of thousands of routers and this allows people all over the globe to communicate using for example, email. Routers by default do not send broadcasts as it is routing data.
If wires are used, each computer is connected by its own wire to the router. Modern wired-only routers designed for the home or small business typically have one “input” port (to the Internet) and four “output” ports, one or more of which can be connected to other computers. A typical modern home wireless router, in addition to having four wired ports, also allows several devices to connect with it wirelessly. Most modern personal computers are built with a wired port (almost always an Ethernet type), which allows them to connect to a router with the addition of just a cable (typically a Category type). To connect with a wireless router, a device must have an adapter. This is sometimes, but not always, included with the computer at manufacture. Some electronic games, including handheld electronic games, have an adapter built-in, or one can be added later.
More technically, a router is a networking device whose software and hardware are usually tailored to the tasks of routing and forwarding information. Routers connect two or more logical subnets, which do not necessarily map one-to-one to the physical interfaces of the router. The term “layer 3 switching” is often used interchangeably with routing, but switch is a general term without a rigorous technical definition. In marketing usage, a switch is generally optimized for Ethernet LAN interfaces and may not have other physical interface types.
In comparison, the network hub (predecessor of the “switch” or “switching hub”) does not do any routing, instead every packet it receives on one network line gets forwarded to all the other network lines.
Routers operate in two different planes:
Control plane, in which the router learns the outgoing interface that is most appropriate for forwarding specific packets to specific destinations,
Forwarding plane, which is responsible for the actual process of sending a packet received on a logical interface to an outbound logical interface.
Types of routers
Routers may provide connectivity inside enterprises, between enterprises and the Internet, and inside Internet Service Providers (ISPs). The largest routers (for example the CiscoCRS-1 or Juniper T1600) interconnect ISPs, are used inside ISPs, or may be used in very large enterprise networks. The smallest routers provide connectivity for small and home offices.
Routers for Internet connectivity and internal use
Router Cisco Wireless
Routers intended for ISP and major enterprise connectivity will almost invariably exchange routing information with the Border Gateway Protocol (BGP). RFC 4098 defines several types of BGP-speaking routers:
Edge Router – Placed at the edge of an ISP network, it speaks external BGP (EBGP) to a BGP speaker in another provider or large enterprise Autonomous System(AS).
Subscriber Edge Router – Located at the edge of the subscriber’s network, it speaks EBGP to its provider’s AS(s). It belongs to an end user (enterprise) organization.
Inter-provider Border Router – Interconnecting ISPs, this is a BGP speaking router that maintains BGP sessions with other BGP speaking routers in other providers’ ASes.
Core router – A router that resides within the middle or backbone of the LAN network rather than at its periphery.
Within an ISP – Internal to the provider’s AS, such a router speaks internal BGP (IBGP) to that provider’s edge routers, other intra-provider core routers, or the provider’s inter-provider border routers.
“Internet backbone” – The Internet does not have a clearly identifiable backbone, as did its predecessors. See default-free zone(DFZ). Nevertheless, it is the major ISPs’ routers that make up what many would consider the core. These ISPs operate all four types of the BGP-speaking routers described here. In ISP usage, a “core” router is internal to an ISP, and used to interconnect its edge and border routers. Core routers may also have specialized functions in virtual private networks based on a combination of BGP and Multi-Protocol Label Switching (MPLS).
Routers are also used for port forwarding for private servers.
Describe the purpose and functions of various network devices 2
Switch
Look at a switch as a device that allows PCs to connect to it. For example, a switch could have say 24 connections which will allow 24 PCs to connect to it. Each PC would connect to what is called a switch-port on the switch using a (Ethernet) cable. Now look visualize an office environment on one floor. Say there are 16 people on the office floor and each person’s PC connects to a switch-port. The switch setup in this way allow the sixteen people to communicate with each other. You can also connect a printer to one of the switch-ports thereby enabling everyone to access to print documents.
You can also connect switches to each other using a (crossover Ethernet) cable. So for example, a office block had 6 floors with 20 people each. Each switch on the floor could connect to each other (trunking) thereby allowing everybody in the building to communicate with each other. Switches traditionally operates at Layer2 of the OSI Layer (Data-link layer) but is common these days to see hybrid ones with Layer 3 functionality. Where routers prevent broadcasts, switches allow broadcasting of packets so that every switch-ports can learn if data is destined for them. You can contain broadcasts by turning on Layer3 features using VLANs. A switch stores the MAC Address of every device which is connected to it.
The switch will then evaluate every frame that passes through it. The switch will examine the destination MAC Address in each frame. Based upon the destination MAC Address, the switch will then decide which port to copy the frame to. If the switch does not recognize the MAC Address, it will not know which port to send the frame to and broadcasts all ports. If the switch does not have layer3 options then it can connect to a router to prevent broadcasts and route packets to different networks.
Routers and switches vary in size and performance to meet differing needs. High performance switches allow modules to be connected or inserted to enable routing, security services and voice over IP (VOIP).
Function
The network switch, packet switch (or just switch) plays an integral part in most Ethernet local area networks or LANs. Mid-to-large sized LANs contain a number of linked managed switches. Small office/home office (SOHO) applications typically use a single switch, or an all-purpose converged device such as gateway access to small office/home broadband services such as DSL router or cable Wi-Fi router. In most of these cases, the end user device contains a router and components that interface to the particular physical broadband technology, as in the Linksys 8-port and 48-port devices. User devices may also include a telephone interface to VoIP.
In the context of a standard 10/100 Ethernet switch, a switch operates at the data-link layer of the OSI model to create a different collision domain per switch port. If you have 4 computers A/B/C/D on 4 switch ports, then A and B can transfer data between them as well as C and D at the same time, and they will never interfere with each others’ conversations. In the case of a “hub” then they would all have to share the bandwidth and run in Half duplex. The result is that there would be collisions and retransmissions. Using a switch is called micro-segmentation. It allows you to have dedicated bandwidth on point to point connections with every computer and to therefore run in Full with no collisions.
Role of Switches in networks
Switches may operate at one or more OSI layers, including physical, data link, network, or transport (i.e., end-to-end). A device that operates simultaneously at more than one of these layers is known as a multilayer switch.
In switches intended for commercial use, built-in or modular interfaces make it possible to connect different types of networks, including Ethernet, Fibre Channel, ATM, ITU-T G.hn and 802.11. This connectivity can be at any of the layers mentioned. While Layer 2 functionality is adequate for speed-shifting within one technology, interconnecting technologies such as Ethernet and token ring are easier at Layer 3.
Interconnection of different Layer 3 networks is done by routers. If there are any features that characterize “Layer-3 switches” as opposed to general-purpose routers, it tends to be that they are optimized, in larger switches, for high-density Ethernet connectivity.
In some service provider and other environments where there is a need for a great deal of analysis of network performance and security, switches may be connected between WAN routers as places for analytic modules. Some vendors provide firewall, network intrusion detection, and performance analysis modules that can plug into switch ports. Some of these functions may be on combined modules.
In other cases, the switch is used to create a mirror image of data that can go to an external device. Since most switch port mirroring provides only one mirrored stream, network hubs can be useful for fanning out data to several read-only analyzers, such as intrusion detection systems and packet sniffers.
Describe the purpose and functions of various network devices 3
While switches may learn about topologies at many layers, and forward at one or more layers, they do tend to have common features. Other than for high-performance applications, modern commercial switches use primarily Ethernet interfaces, which can have different input and output speeds of 10, 100, 1000 or 10,000 megabits per second. Switch ports almost always default to Full duplex operation, unless there is a requirement for interoperability with devices that are strictly Half duplex. Half duplex means that the device can only send or receive at any given time, whereas Full duplex can send and receive at the same time.
At any layer, a modern switch may implement power over Ethernet (PoE), which avoids the need for attached devices, such as an IP telephone or wireless access point, to have a separate power supply. Since switches can have redundant power circuits connected to uninterruptible, the connected device can continue operating even when regular office power fails.
Hubs versus higher-layer switches
Layer-1
A network hub, or repeater, is a fairly unsophisticated network device. Hubs do not manage any of the traffic that comes through them. Any packet entering a port is broadcast out or “repeated” on every other port, except for the port of entry. Since every packet is repeated on every other port, packet collisions result, which slows down the network.
There are specialized applications where a hub can be useful, such as copying traffic to multiple network sensors. High end switches have a feature which does the same thing called port mirroring. There is no longer any significant price difference between a hub and a low-end switch.
Layer 2
A network bridge, operating at the Media Access Control (MAC) sub-layer of the data link layer, may interconnect a small number of devices in a home or office. This is a trivial case of bridging, in which the bridge learns the MAC address of each connected device. Single bridges also can provide extremely high performance in specialized applications such as storage area networks.
Classic bridges may also interconnect using a spanning tree protocol that disables links so that the resulting local area network is a tree without loops. In contrast to routers, spanning tree bridges must have topologies with only one active path between two points. The older IEEE 802.1D spanning tree protocol could be quite slow, with forwarding stopping for 30 seconds while the spanning tree would re-converge. A Rapid Spanning Tree Protocol was introduced as IEEE 802.1w, but the newest edition of IEEE 802.1D-2004, adopts the 802.1w extensions as the base standard. The IETF is specifying the TRILL protocol, which is the application of link-state routing technology to the layer-2 bridging problem. Devices which implement TRILL, called Routing Bridges, combine the best features of both routers and bridges.
While “layer 2 switch” remains more of a marketing term than a technical term, the products that were introduced as “switches” tended to use micro segmentation and Full duplex to prevent collisions among devices connected to Ethernets. By using an internal forwarding plane much faster than any interface, they give the impression of simultaneous paths among multiple devices.
Once a bridge learns the topology through a spanning tree protocol, it forwards data link layer frames using a layer 2 forwarding method. There are four forwarding methods a bridge can use, of which the second through fourth method were performance-increasing methods when used on “switch” products with the same input and output port speeds:
Store and forward – The switch buffers and, typically, performs a checksum on each frame before forwarding it on.
Cut through – The switch reads only up to the frame’s hardware address before starting to forward it. There is no error checking with this method.
Fragment free – A method that attempts to retain the benefits of both “store and forward” and “cut through”. Fragment free checks the first 64 bytes of the frame, where addressing information is stored. According to Ethernet specifications, collisions should be detected during the first 64 bytes of the frame, so frames that are in error because of a collision will not be forwarded. This way the frame will always reach its intended destination. Error checking of the actual data in the packet is left for the end device in Layer 3 or Layer 4 (OSI), typically a router.
Adaptive switching – A method of automatically switching between the other three modes.
Cut-through switches have to fall back to store and forward if the outgoing port is busy at the time the packet arrives. While there are specialized applications, such as storage area networks, where the input and output interfaces are the same speed, this is rarely the case in general LAN applications. In LANs, a switch used for end user access typically concentrates lower speed (e.g., 10/100 Mbit/s) into a higher speed (at least 1 Gbit/s). Alternatively, a switch that provides access to server ports usually connects to them at a much higher speed than is used by end user devices.
Layer 3
Within the confines of the Ethernet physical layer, a layer 3 switch can perform some or all of the functions normally performed by a router. A true router is able to forward traffic from one type of network connection (e.g., T1, DSL) to another (e.g., Ethernet, Wi-Fi).
The most common layer-3 capability is awareness of IP multicast. With this awareness, a layer-3 switch can increase efficiency by delivering the traffic of a multicast group only to ports where the attached device has signalled that it wants to listen to that group. If a switch is not aware of multicasting and broadcasting, frames are also forwarded on all ports of each broadcast domain, but in the case of IP multicast this causes inefficient use of bandwidth. To work around this problem some switches implement IGMP snooping.
Layer 4
While the exact meaning of the term Layer-4 switch is vendor-dependent, it almost always starts with a capability for network address translation, but then adds some type of load distribution based on TCP sessions.
The device may include a state full firewall, a VPN concentrator, or be an IPSec security gateway.
Layer 7
Layer 7 switches may distribute loads based on URL or by some installation-specific technique to recognize application-level transactions. A Layer-7 switch may include a web cache and participate in a content delivery network.
Types of switches
Form factor
Switchport
Desktop, not mounted in an enclosure, typically intended to be used in a home or office environment outside of a wiring closet
Rack mounted
Chassis – with swappable “switch module” cards. e.g. Alcatel’s Omni Switch 8000; Cisco Catalyst switch 4500 and 6500; 3Com 7700, 7900E, 8800.
Unmanaged switches – These switches have no configuration interface or options. They are plug and play. They are typically the least expensive switches, found in home, SOHO, or small businesses. They can be desktop or rack mounted.
Configuration options
Managed switches – These switches have one or more methods to modify the operation of the switch. Common management methods include: a serial console or command line interface accessed via telnet or Secure Shell, an embedded Simple Network Management Protocol (SNMP) agent allowing management from a remote console or management station, or a web interface for management from a web browser. Examples of configuration changes that one can do from a managed switch include: enable features such as Spanning Tree Protocol, set port speed, create or modify Virtual LANs (VLANs), etc.
Two sub-classes of managed switches are marketed today:
Smart (or intelligent) switches – These are managed switches with a limited set of management features. Likewise “web-managed” switches are switches which fall in a market niche between unmanaged and managed. For a price much lower than a fully managed switch they provide a web interface (and usually no CLI access) and allow configuration of basic settings, such as VLANs, port-speed and duplex.
Enterprise Managed (or fully managed) switches – These have a full set of management features, including Command Line Interface, SNMP agent, and web interface. They may have additional features to manipulate configurations, such as the ability to display, modify, backup and restore configurations. Compared with smart switches, enterprise switches have more features that can be customized or optimized, and are generally more expensive than “smart” switches. Enterprise switches are typically found in networks with larger number of switches and connections, where centralized management is a significant savings in administrative time and effort. A stackable switch is a version of enterprise-managed switch.
Traffic monitoring on a switched network
Switches
Unless port mirroring or other methods such as RMON or SMON are implemented in a switch, it is difficult to monitor traffic that is bridged using a switch because all ports are isolated until one transmits data, and even then only the sending and receiving ports can see the traffic. These monitoring features rarely are present on consumer-grade switches.
Two popular methods that are specifically designed to allow a network analyst to monitor traffic are:
Port mirroring – the switch sends a copy of network packets to a monitoring network connection.
SMON - “Switch Monitoring” is described by RFC 2613 and is a protocol for controlling facilities such as port mirroring.
Another method to monitor may be to connect a Layer-1 hub between the monitored device and its switch port. This will induce minor delay, but will provide multiple interfaces that can be used to monitor the individual switch port.
Typical switch management features
Turn some particular port range on or off
Link speed and duplex settings
Priority settings for ports
MAC filtering and other types of “port security” features which prevent MAC flooding
Use of Spanning Tree Protocol
SNMP monitoring of device and link health
Port mirroring (also known as: port monitoring, spanning port, SPAN port, roving analysis port or link mode port)
Link aggregation (also known as bonding, trunking or teaming)
VLAN settings
802.1X network access control
IGMP snooping
Link aggregation allows the use of multiple ports for the same connection achieving higher data transfer speeds. Creating VLANs can serve security and performance goals by reducing the size of the broadcast domain.
Select the components required to meet a given network specification 1
Computer networks can be classified according to the hardware and software technology that is used to interconnect the individual devices in the network, such as optical fibre, Ethernet, Wireless LAN, HomePNA, Power line communication or G.hn.
Ethernet uses physical wiring to connect devices. Frequently deployed devices include hubs, switches, bridges and/or routers. Wireless LAN technology is designed to connect devices without wiring. These devices use radio waves or infrared signals as a transmission medium. ITU-T G.hn technology uses existing home wiring (coaxial cable and power lines) to create a high-speed (up to 1 Gigabit/s) local area network.
Wired technologies
Unshielded Twisted Pair (UTP) Cable
UTP Patch cable
Twisted pair cabling comes in two varieties: shielded and unshielded. Unshielded twisted pair (UTP) is the most popular and is generally the best option for school networks (See figure)
The quality of UTP may vary from telephone-grade wire to extremely high-speed cable. The cable has four pairs of wires inside the jacket. Each pair is twisted with a different number of twists per inch to help eliminate interference from adjacent pairs and other electrical devices. The tighter the twisting, the higher the supported transmission rate and the greater the cost per foot. The EIA/TIA (Electronic Industry Association/Telecommunication Industry Association) has established standards of UTP and rated six categories of wire (additional categories are emerging).
ANSI TIA/EIA 568-B2 - Twisted Pair Categories of Cable Standards
Describes cable standards Level 1 through Cat5e - *** do not confuse this with the 568-B standard, which describes RJ45 wiring pinouts, and colour schemes
In the mid 1980s, companies representing the telecommunications and computer industries were concerned with the lack of a standard for building telecommunications cabling systems. In response to that concern, the Electronic Industries Association (EIA) developed a standard called TIA/EIA-568-A. The higher the standard, the more twists per inch the pairs have, and the better the quality of the jack (RJ-45 typically).
ANSI/EIA (American National Standards Institute/Electronic Industries Association) Standard 568 is one of several standards that specify “categories” (the singular is commonly referred to as “CAT”) of twisted pair cabling systems (wires, junctions, and connectors) in terms of the data rates that they can sustain. The specifications describe the cable material as well as the types of connectors and junction blocks to be used in order to conform to a category. These categories are:
ANSI Category Maximum data rate Usual application
CAT 1 Up to 1 Mbps (1 MHz ) analogue voice (POTS)
Integrated Services Digital Network Basic Rate Interface in ISDN
Doorbell wiring
CAT 2 4 Mbps Mainly used in the IBM Cabling System for Token Ring networks
CAT 3 16 Mbps Voice and data on 10BASE-T Ethernet
CAT 4 20 Mbps Used in 16 Mbps Token Ring
Otherwise not used much
CAT 5 100 Mbps
1000 Mbps (4 pair) 100 Mbps TPDDI
155 Mbps ATM
CAT 5E
(ISO Class D) 100 Mbps TPDDI 155 Mbps ATM
CAT 6
(ISO Class E) Up to 400 MHz Super-fast broadband applications (proposed standard). Used with GigE (1000 Mbps or 1 Gbps)
CAT 7
(ISO Class F) 600-700 MHz Even faster broadband applications (proposed standard)
Note: There is also a standard called 100Base-T4, intended for fast-Ethernet over non-Cat.5 cable. This one uses all four pairs. It is not very common and not supported by nowadays “standard” hardware. HP invented yet another thing called 100Base-VG “AnyLan”, which also uses all four pairs. It’s not very common either.
While longer connections for Gigabit Ethernet use optical fibre, the goal is to leverage the CAT 5 twisted-pair wiring most organizations already have in place for connections out to the desktop. (Four pairs of twisted pair are used.)
include connections to third parties.
Select the components required to meet a given network specification 2
The CAT 6 specification is able to support GbE (GigE, or Gigabit Ethernet) which runs at 1000 Mbps, or 1 Gbps. GigE uses fibre typically but can also use Cat6.
Category 1 (Cat1 – Level 1)
This category consists of basic telecommunications and power-limited circuit cables. There are no electrical performance test or bandwidth requirements for this classification. Level 1 cabling may not be used in horizontal cabling systems.
Category 2 (Cat2 – Level 2)
This category consists of cables specified to 1 MHz Level 2 cabling may not be used in horizontal wiring cabling systems.
Category 3 (Cat 3)
3-4 twists per foot of cable. This is a performance designation for twisted-pair cable and connecting hardware that can support frequency transmission up to 16 MHz, and data rates of 10 Mbps. Category 3 has the capability to support low speed data applications, performing to the acceptable minimum for 100 ohm cabling systems; however it is now primarily used for telephone wiring.
Category 4 (rarely used)
This category consists of cables and connectors specified up to 20 MHz and data rates of 16 Mbps. Since the development of Category 5, however, Category 4 wiring systems are rarely used.
Category 5 (Cat 5)
3-4 twists per inch of cable (12 times more twists than Cat 3). This category consists of cables and connectors specified up to 100 MHz and data rates of 100 Mbps, providing optimal performance for all data and phone systems. These systems are quickly becoming the standard because they provide a “safety net” to help ensure that current and future high-speed applications will run with peak accuracy, efficiency and throughput.
Ethernet cable standard defined by the Electronic Industries Association and Telecommunications Industry Association (commonly known as EIA/TIA). CAT5 is the 5th generation of twisted pair Ethernet cabling and the most popular of all twisted pair cables in use today.
Cat5 cable contains four pairs of copper wire. CAT5 supports Fast (100 Mbps) Ethernet and comparable alternatives such as ATM. As with all other types of twisted pair EIA/TIA cabling, CAT5 cable runs are limited to a maximum recommended run rate of 100m (328 feet).
Although CAT5 cable usually contains four pairs of copper wire, Fast Ethernet communications only utilize two pairs.
Solid and Stranded Cat5 Cable - Twisted pair cable like CAT5 comes in two main varieties, solid and stranded. Solid Cat5 cable supports longer runs and works best in fixed wiring configurations like office buildings. Stranded Cat5 cable, on the other hand, is more pliable and better suited for shorter-distance, movable cabling such as on-the-fly “patch” cabling.
Though newer cable technologies like CAT6 and CAT7 are being developed, CAT5 cable remains the popular choice, because it is both affordable and plenty fast enough for today’s LANs.
Category 5 Enhanced (Cat 5E )
More twists per inch than Cat5 – supports short-run Gigabit Ethernet (1000 Mbps) networking by utilizing all four wire pairs and is backward-compatible with ordinary CAT5.
Category 6 (Cat 6)
Cat6 is an Ethernet cable standard defined by the Electronic Industries Association and Telecommunications Industry Association (commonly known as EIA/TIA). CAT6 is the 6th generation of twisted pair Ethernet cabling.
Cat6 cable contains four pairs of copper wire. Unlike CAT5 but like Cat5e, Cat6 utilizes all four pairs. CAT6 supports Gigabit (1000 Mbps) Ethernet and supports communications at more than twice the speed of CAT5e, the other popular standard for Gigabit Ethernet cabling.
As with all other types of twisted pair EIA/TIA cabling, CAT6 cable runs are limited to a maximum recommended run rate of 100m (328 feet).
Twisted pair cable like CAT6 comes in two main varieties, solid and stranded. Solid CAT6 cable supports longer runs and works best in fixed wiring configurations like office buildings. Stranded CAT6 cable, on the other hand, is more pliable and better suited for shorter-distance, movable cabling such as “patch” cables.
CAT6 comes at a significantly higher price tag than CAT5 or CAT5e, and today’s applications simply can’t take advantage of CAT6’s better performance. However, if wiring a home or building for the long term, one may still consider using CAT6.
Category 7 (Cat 7)
This category is still in the development stage and will use a braided shield surrounding all four foil shielded pairs to reduce noise and interference. The connector is RJ45, delivered only through pins 1/2 and 7/8.
The cable is ISO Class F, and is defined in IS11801 2ndEdition, 2002.
Performance is characterized to 600MHz and Crosstalk isolation is >20dB more than Class D cables (cat5e). Each of the conductor pairs has its own shielding, in addition to the outer shielding.
Initially, the default Cat7 connector was RJ45, but several new connector designs came out to challenge this.
Several new “Cat. 7″ connectors are under Development. (e.g.: AMP, BKS, Siemon, Telesafe, T&B) z IEC/TC48 is currently discussing new standard proposals.
The Cat7 Tera Connector - the IEC vote of 18 countries represents a significant achievement for structured cabling. For the first time, a non-RJ-style connector interface has been internationally standardised for four pair connections in a structured cabling system. During the interface selection conducted by ISO/IEC, an independent panel was asked to judge six different non-RJ-style connector proposals. Based on 48 separate criteria, including size, complexity, manufacturability, user-friendliness and transmission performance, the TERA™ interface was ranked the best overall choice for delivering the demanding bandwidth specified in the standard.
Finally, the Siemon “Tera” connector was chosen by the IEC as the official Cat7 connector and is described in IEC 61076-3-104. It is now the standard interface for Category 7/Class F and broadcast communications technology (BCT) cabling and is commercially available.
“TERA™ system offers enhanced performance to 1.2 GHz that constitutes an electrical superset of all other balanced cabling categories and classes. It has the unique ability to replace other high-speed media such as 50- and 75-ohm coaxial media used for broadband video, as well as 150-ohm Shielded Twisted Pair (STP). Its innovative design allows for up to four individual 1.2 GHz balanced pair connections in the same space as a single keystone jack permitting users to integrate video, voice and data services over a single cabling link. The outstanding pair isolation that makes this capability possible provides for unsurpassed alien crosstalk performance – an important consideration for emerging applications like 10GBASE-T. It is positioned to revolutionize structured cabling systems both for commercial and residential applications.”
In addition to this ISO/IEC approval , the Siemon TERA™ connector interface has been chosen by ISO/IEC as the primary interface for the Broadcast and Communications technology (BCT) and Small Office Home Office (SOHO) standard being developed by draft standard ISO/IEC 15018 (ISO/IEC JTC 1/SC 25 N822)
Siemon offers a complete line of TERA™ products including cable, outlets, patch panels, modular patch cords and more. TERA™ is a fully shielded (S/FTP) cabling solution that virtually eliminates emissions and susceptibility to electromagnetic interference.
Select the components required to meet a given network specification 3
Fiber Optic
Fibre optic cabling consists of a centre glass core surrounded by several layers of protective materials (See fig. 5). It transmits light rather than electronic signals eliminating the problem of electrical interference. This makes it ideal for certain environments that contain a large amount of electrical interference. It has also made it the standard for connecting networks between
Fiber optic cable has the ability to transmit signals over much longer distances than coaxial and twisted pair. It also has the capability to carry information at vastly greater speeds. This capacity broadens communication possibilities to include services such as video conferencing and interactive services. The cost of fiber optic cabling is comparable to copper cabling; however, it is
The centre core of fiber cables is made from glass or plastic fibers (see fig). A plastic coating then cushions the fiber centre, and Kevlar fibers help to strengthen the cables and prevent breakage. The outer insulating jacket made of Teflon or PVC.
Fiber optic cable
There are two common types of fiber cables — single mode and multimode. Multimode cable has a larger diameter; however, both cables provide high bandwidth at high speeds. Single mode can provide more distance, but it is more expensive.
Ethernet Cable Summary
Specification Cable Type
10BaseT Unshielded Twisted Pair
10Base2 Thin Coaxial
10Base5 Thick Coaxial
100BaseT Unshielded Twisted Pair
100BaseFX Fiber Optic
100BaseBX Single mode Fiber
100BaseSX Multimode Fiber
1000BaseT Unshielded Twisted Pair
1000BaseFX Fiber Optic
1000BaseBX Single mode Fiber
1000BaseSX Multimode Fiber
Wireless LANs
More and more networks are operating without cables, in the wireless mode. Wireless LANs use high frequency radio signals, infrared light beams, or lasers to communicate between the workstations and the file server or hubs. Each workstation and file server on a wireless network has some sort of transceiver/antenna to send and receive the data. Information is relayed between transceivers as if they were physically connected. For longer distance, wireless communications can also take place through cellular telephone technology, microwave transmission, or by satellite.
Wireless networks are great for allowing laptop computers or remote computers to connect to the LAN. Wireless networks are also beneficial in older buildings where it may be difficult or impossible to install cables.
The two most common types of infrared communications used in schools are line-of-sight and scattered broadcast. Line-of-sight communication means that there must be an unblocked direct line between the workstation and the transceiver. If a person walks within the line-of-sight while there is a transmission, the information would need to be sent again. This kind of obstruction can slow down the wireless network. Scattered infrared communication is a broadcast of infrared transmissions sent out in multiple directions that bounces off walls and ceilings until it eventually hits the receiver. Networking communications with laser are virtually the same as line-of-sight infrared networks.
Wireless standards and speeds
The Wi-Fi Alliance is a global, non-profit organization that helps to ensure standards and interoperability for wireless networks, and wireless networks are often referred to as Wi-Fi (Wireless Fidelity). The original Wi-Fi standard (IEEE 802.11) was adopted in 1997. Since then many variations have emerged (and will continue to emerge). Wi-Fi networks use the Ethernet protocol.
Standard Max Speed Typical Range
802.11a 54 Mbps 150 feet
802.11b 11 Mbps 300 feet
802.11g 54 Mbps 300 feet
Advantages of wireless networks:
Wireless Lan
Mobility – With a laptop computer or mobile device, access can be available throughout a school, at the mall, on an airplane, etc. More and more businesses are also offering free Wi-Fi access.
Fast setup – If your computer has a wireless adapter, locating a wireless network can be as simple as clicking “Connect to a Network” — in some cases, you will connect automatically to networks within range.
Cost – Setting up a wireless network can be much more cost effective than buying and installing cables.
Expandability – Adding new computers to a wireless network is as easy as turning the computer on (as long as you do not exceed the maximum number of devices).
Disadvantages of wireless networks:
Security
Security – Wireless networks are much more susceptible to unauthorized use. If you set up a wireless network, be sure to include maximum security. You should always enable WEP (Wired Equivalent Privacy) or WPA (Wi-Fi Protected Access), which will improve security and help to prevent virtual intruders and freeloaders.
Interference – Because wireless networks use radio signals and similar techniques for transmission, they are susceptible to interference from lights and electronic devices.
Inconsistent connections – How many times have you hears “Wait a minute, I just lost my connection?” Because of the interference caused by electrical devices and/or items blocking the path of transmission, wireless connections are not nearly as stable as those through a dedicated cable.
Power consumption – The wireless transmitter in a laptop requires a significant amount of power; therefore, the battery life of laptops can be adversely impacted. If you are planning a laptop project in your classroom, be sure to have power plugs and/or additional batteries available.
Speed – The transmission speed of wireless networks is improving; however, faster options (such as gigabit Ethernet) are available via cables. In addition, if set up a wireless network at home, and you are connecting to the Internet via a DSL modem (at perhaps 3 Mbps), your wireless access to the Internet will have a maximum of 3 Mbps connection speed.
Scale
Networks are often classified as local area network (LAN), wide area network (WAN), metropolitan area network (MAN), personal area network (PAN), virtual private network (VPN), campus area network (CAN), storage area network (SAN), and others, depending on their scale, scope and purpose. Usage, trust level, and access right often differ between these types of network. For example, LANs tend to be designed for internal use by an organization’s internal systems and employees in individual physical locations (such as a building), while WANs may connect physically separate parts of an organization and may
Use the OSI and TCP/IP models and their associated protocols to explain how data flows in a network 1
OSI Model
7. Application layer
6. Presentation layer
5. Session layer
4. Transport layer
3. Network layer
2. Data Link layer
1. Physical layer
TCP/IP (DARPA) Model
4. Application layer
3. Transport layer
2. Network layer (Internet layer)
1. Link layer (Network Interface layer)
The OSI Network Model
The OSI (Open System Interconnection) 7-layer reference model defines a concept of moving information between networked computers. It describes how information flows from one end-user application through a network into another application. This model is considered the primary architectural model for inter-computer communication. Each of the 7 OSI layers are reasonably self-contained, and handle a separate group of tasks.
Layer 1 - Physical - The Physical layer defines the electrical, mechanical, and all physical hardware means of sending and receiving data itself. It includes cables, cards, and all physical aspects. It conveys the bit stream through the network at the electrical and mechanical level. The physical layer specifications also define characteristics, such as voltage levels, timing of impulses, physical data rates, max transmission distance, and physical connectors. Physical layer implementations can be categorized as either WAN or LAN specifications.
Layer 2 - Data Link - The data link layer defines the format of data, and provides its reliable transit across the physical network link. At this layer, bits are encoded/decoded into data packets (with protocol-specific headers, including checksums, source/destination addresses, etc.). The layer facilitates transmission protocol management, flow control, frame synchronization, and handles any errors in the physical layer. It contains two sub-layers - MAC (Media Access Control), and LLC (Logical Link Control).
The IEEE MAC specification defines MAC addresses, which enable multiple devices to uniquely identify each other at the data link layer. The MAC sub-layer manages protocol access to the physical network medium. It controls how a network device gains access to, and permission to transmit data.
LLC manages communications between devices over a single link of a network. It controls frame synchronization, flow control and error checking.
Layer 3 - Network - The Network layer provides network addressing (which differs from the data link layer MAC address). It also facilitates switching, routing, error handling, congestion control, and packet sequencing. It allows for defining the logical network layout, and virtual logical paths for transmitting data between network nodes. The Internetwork Protocol (IP) operates at this layer. IP defines network addresses in a way that route selection can be determined systematically by comparing the source network address with the destination address, and applying the subnet mask. Routers operate at this layer to determine how to forward packets. Most of the design and configuration of network layout is at the networking layer.
Layer 4 - Transport - The Transport layer segments data (into packets) for transport across the network. It ensures complete data transfer by providing flow control, multiplexing, error checking and error recovery (retransmissions). Flow control manages data transmission between devices, so that the transmitting device does not send more data than the receiving device can process. Multiplexing allows for data from different applications to be transmitted through a single physical link. Such virtual “circuits” are established, maintained and terminated by the transport layer.
The most common transport layer protocols are TCP (Transmission Control Protocol) and UDP (User Datagram Protocol), with reliability and speed being the main difference between them.
Layer 5 - Session - The Session layer generally deals with session and connection coordination. It establishes, manages and terminates communications sessions. Sessions consist of service requests and responses that occur between applications in different network devices. Session protocol implementations include RPC (Remote Procedure Call), ZIP (Zone Information Protocol), AppleTalk, SCP (Session Control Protocol).
Layer 6 – Presentation - The Presentation layer deals with conversion and coding of data from application to network format. It ensures the application layer data can be readable by other systems’ application layer. This layer formats and encrypts data, providing compatibility between systems. It is sometimes called the “syntax layer”.
Layer 7 - Application - The Application layer supports and interacts directly with software applications. Its functions include identifying communication partners, determining resource availability and synchronizing communication. Some examples of application layer implementations include FTP (File Transfer Protocol), Telnet, SMTP (Simple Mail Transfer Protocol).
Use the OSI and TCP/IP models and their associated protocols to explain how data flows in a network 2
Layer # Name Mnemonic Encapsulation Units Devices or Components Keywords/Description
7 Application All data PC Network services for application processes, such as file, print, messaging, database services
6 Presentation People data Standard interface to data for the application layer. MIME encoding, data encryption, conversion, formatting, compression
5 Session Seem data Inter-host communication. Establishes, manages and terminates connection between applications
4 Transport To segments End-to-end connections and reliability. Segmentation/de-segmentation of data in proper sequence. Flow control
3 Network Need packets router Logical addressing and path determination. Routing. Reporting delivery errors
2 Data Link Data frames bridge, switch, NIC Physical addressing and access to media. Two sub-layers: Logical Link Control (LLC) and Media Access Control (MAC)
1 Physical Processing bits repeater, hub, tranciever Binary transmission signals and encoding. Layout of pins, voltages, cable specifications, modulation
The TCP/IP Network Model
The TCP/IP protocol suite forms the basis of the Internet. It is the most widely used form of networking between computers. TCP/IP is a combination of protocols at different layers that is designed around simple 4-layer scheme. It combines/splits some adjacent OSI layers, and omits some features. The 4-layer TCP/IP model is also known as the DARPA model, named after the U.S. government agency that initially developed TCP/IP. The 4 layers are as follows:
Layer 1 – Link – The Link layer defines the device driver and network hardware (network interface card).
Layer 2 - Network – The Network layer handles basic communication, addressing and routing. IP, ICMP, ARP and IGMP protocols are at the network layer.
Layer 3 - Transport – The Transport layer handles flow of data among applications. It segments data into packets for transport over the network. TCP and UDP operate at the transport layer.
Layer 4 - Application – The Application layer handles details of the particular end-user applications. Commonly used TCP/IP applications include Telnet, FTP, SMTP, SNMP, DNS, RIP, NFS, NTP, Trace-route.
TCP/IP Protocol Stack
TCP/IP Layer TCP/IP Protocols
4 Application data FTP, HTTP, POP3, IMAP, telnet, SMTP, DNS, TFTP
3 Transport segments TCP, UDP, IGMP, ICMP
2 Internet packets IP – IPSEC
1 Network Access/ Interface frames Ethernet – Token Ring – Frame Relay – ATM
bits (Hardware)
SIMILARITIES
The main similarities between the two models are:
Both of the models share a similar architecture. This can be illustrated by the fact that both of them are constructed with layers.
Both of the models share a common “application layer”. However in practice this layer includes different services depending upon each model.
Both models have comparable transport and network layers.- This can be illustrated by the fact that whatever functions are performed between the presentation and network layer of the OSI model similar functions are performed at the Transport layer of the TCP/IP model.
Knowledge of both models is required by networking professionals.
Both models assume that packets are switched.- Basically this means that individual packets may take differing paths in order to reach the same destination.
DIFFERENCES
The main differences between the two models are:
TCP/IP Protocols are considered to be standards around which the internet has developed. The OSI model however is a ”generic, protocol – independent standard.”
TCP/IP combines the presentation and session layer issues into its application layer.
TCP/IP combines the OSI data link and physical layers into the network access layer.
TCP/IP appears to be a more simpler model and this is mainly due to the fact that it has fewer layers.
TCP/IP is considered to be a more credible model- This is mainly due to the fact because TCP/IP protocols are the standards around which the internet was developed therefore it mainly gains creditability due to this reason. Where as in contrast networks are not usually built around the OSI model as it is merely used as a guidance tool.
The OSI model consists of 7 architectural layers whereas the TCP/IP only has 4 layers.
Describe common networking applications including web applications
Application Layer protocols generally treat the transport layer (and lower) protocols as “black boxes” which provide a stable network connection across which to communicate, although the applications are usually aware of key qualities of the transport layer connection such as the end point IP addresses and port numbers. As noted above, layers are not necessarily clearly defined in the Internet protocol suite. Application layer protocols are most often associated with client-server applications, and the commoner servers have specific ports assigned to them by the IANA: HTTP has port 80; Telnet has port 23; etc. Clients, on the other hand, tend to use ephemeral ports, i.e. port numbers assigned at random from a range set aside for the purpose.
Transport and lower level layers are largely unconcerned with the specifics of application layer protocols. Routers and switches do not typically “look inside” the encapsulated traffic to see what kind of application protocol it represents, rather they just provide a conduit for it. However, some firewall and bandwidth throttling applications do try to determine what’s inside, as with the Resource Reservation Protocol (RSVP). It’s also sometimes necessary for Network Address Translation (NAT) facilities to take account of the needs of particular application layer protocols. (NAT allows hosts on private networks to communicate with the outside world via a single visible IP address using port forwarding, and is an almost ubiquitous feature of modern domestic broadband routers).
DNS (Domain Name System)
DHCP (Dynamic Host Configuration Protocol)
FTP (File Transfer Protocol)
HTTP (HyperText Transfer Protocol)
IMAP (Internet Message Access Protocol)
IRC (Internet Relay Chat)
POP3 (Post Office Protocol Version 3)
SSH (Secure Shell)
SMTP (Simple Mail Transfer Protocol)
Telnet (a remote terminal access protocol)
TFTP (Trivial File Transfer Protocol, a simple file transfer protocol)
web application
In software engineering, a web application is an application that is accessed via a web browser over a network such as the Internet or an intranet. The term may also mean a computer software application that is hosted in a browser-controlled environment (e.g. a Java applet) or coded in a browser-supported language (such as JavaScript, combined with a browser-rendered markup language like HTML) and reliant on a common web browser to render the application executable.
Web applications are popular due to the ubiquity of web browsers, and the convenience of using a web browser as a client, sometimes called a thin client. The ability to update and maintain web applications without distributing and installing software on potentially thousands of client computers is a key reason for their popularity, as is the inherent support for cross-platform compatibility. Common web applications include webmail, online retail sales, online auctions, wikis and many other functions.
Describe the purpose and basic operation of the protocols in the OSI and TCP models 1
Protocol layering is a common technique to simplify networking designs by dividing them into functional layers, and assigning protocols to perform each layer’s task.
For example, it is common to separate the functions of data delivery and connection management into separate layers, and therefore separate protocols. Thus, one protocol is designed to perform data delivery, and another protocol, layered above the first, performs connection management. The data delivery protocol is fairly simple and knows nothing of connection management. The connection management protocol is also fairly simple, since it doesn’t need to concern itself with data delivery.
Protocol layering produces simple protocols, each with a few well-defined tasks. These protocols can then be assembled into a useful whole. Individual protocols can also be removed or replaced as needed for particular applications.
The most important layered protocol designs are the Internet’s original TCP/IP model, and the OSI Seven Layer Model. The modern Internet represents a fusion of both models.
OSI protocols
The Open Systems Interconnection (OSI) protocols are a family of information exchange standards developed jointly by the ISO and the ITU-T starting in 1977. The OSI model describes seven layers of interconnection: the Physical Layer (Layer 1), Data Link Layer (Layer 2), Network Layer (Layer 3), Transport Layer (Layer 4), Session Layer (Layer 5), Presentation Layer (Layer 6), and the Application Layer (Layer 7).
OSI protocols are used to implement various networks. The OSI protocol stack is split into layers for modularity and orthogonal design. The layers form a hierarchy of functionality starting with the physical hardware components to the user interfaces at the software application level. Each layer receives information from the layer above, processes it and passes it down to the next layer.
Each layer adds its own encapsulation information (header) to the incoming information before it is passed to the lower layer. Headers generally include address of destination and source, check sums (for error control), type of protocol used in the current layer, and other options such as flow control options and sequence numbers (used to ensure data is sent in order).
Not all layers are mandatory, it depends on the protocols that are implemented. The Manufacturing Automation Protocol (MAP) user group, focused on real-time control of manufacturing robots of various types, implements layer 1 (physical), a two-sub-layer layer 2 (data link) with LLC Type 3 on top of the medium access layer, and then the Layer 7 Manufacturing Message System on top. Layers 3 to 6 are not present. This high-performance stack is intended just for the robots themselves; the robot controller would load files with a full 7-layer stack with FTAM file transfer on top. Parts of the Signalling System 7 stack are OSI derivatives.
Routing Protocols
A routing protocol is a protocol that specifies how routers communicate with each other, disseminating information that enables them to select routes between any two nodes on a computer network, the choice of the route being done by routing algorithms. Each router has a prior knowledge only of networks attached to it directly. A routing protocol shares this information first among immediate neighbours, and then throughout the network. This way, routers gain knowledge of the topology of the network.
The term routing protocol may refer specifically to one operating at layer three of the OSI model, which similarly disseminates topology information between routers.
Although there are many types of routing protocols, three major classes are in widespread use on IP networks:
Interior gateway routing via link-state routing protocols, such as OSPF and IS-IS
Interior gateway routing via path vector or distance vector protocols, such as RIP, IGRP and EIGRP
Exterior gateway routing. Border gateway protocol is the routing protocol used by the public Internet.
Many routing protocols are defined in documents called RFCs.
The specific characteristics of routing protocols include:
the manner in which they either prevent routing loops from forming or break them up if they do
the manner in which they select preferred routes, using information about hop costs
the time they take to converge
how well they scale up
many other factors
In some cases, routing protocols can themselves run over routed protocols: for example, BGP runs over TCP which runs over IP; care is taken in the implementation of such systems not to create a circular dependency between the routing and routed protocols. That a routing protocol runs over particular transport mechanism does not mean that the routing protocol is of layer (N+1) if the transport mechanism is of layer (N).
Routing protocols, according to the OSI Routing framework, are layer management protocols for the network layer, regardless of their transport mechanism:
IS-IS runs over the data link layer
OSPF, IGRP, and EIGRP run directly over IP; OSPF and EIGRP have their own reliable transmission mechanism while IGRP assumed an unreliable transport
RIP runs over UDP
BGP runs over TCP
Interior routing protocols
Interior Gateway Protocols (IGPs) exchange routing information within a single routing domain. A given autonomous system can contain multiple routing domains, or a set of routing domains can be coordinated without being an Internet-participating autonomous system. Common examples include:
IGRP (Interior Gateway Routing Protocol)
EIGRP (Enhanced Interior Gateway Routing Protocol)
OSPF (Open Shortest Path First)
RIP (Routing Information Protocol)
IS-IS (Intermediate System to Intermediate System)
Note that IGRP, a Cisco proprietary routing protocol, is no longer supported. EIGRP accepts IGRP configuration commands, but the internals of IGRP and EIGRP are completely different.
Describe the purpose and basic operation of the protocols in the OSI and TCP models 2
The Transmission Control Protocol (TCP) is one of the core protocols of the Internet Protocol Suite. TCP is one of the two original components of the suite (the other being Internet Protocol, or IP), so the entire suite is commonly referred to as TCP/IP. Whereas IP handles lower-level transmissions from computer to computer as a message makes its way across the Internet, TCP operates at a higher level, concerned only with the two end systems, for example a Web browser and a Web server. In particular, TCP provides reliable, ordered delivery of a stream of bytes from a program on one computer to another program on another computer. Besides the Web, other common applications of TCP include e-mail and file transfer. Among its other management tasks, TCP controls segment size, flow control, the rate at which data is exchanged, and network traffic congestion.
TCP provides a communication service at an intermediate level between an application program and the Internet Protocol (IP). That is, when an application program desires to send a large chunk of data across the Internet using IP, instead of breaking the data into IP-sized pieces and issuing a series of IP requests, the software can issue a single request to TCP and let TCP handle the IP details.
IP works by exchanging pieces of information called packets. A packet is a sequence of bytes and consists of a header followed by a body. The header describes the packet’s destination and, optionally, the routers to use for forwarding until it arrives at its final destination. The body contains the data which IP is transmitting.
Due to network congestion, traffic load balancing, or other unpredictable network behavior, IP packets can be lost or delivered out of order. TCP detects these problems, requests retransmission of lost packets, rearranges out-of-order packets, and even helps minimize network congestion to reduce the occurrence of the other problems. Once the TCP receiver has finally reassembled a perfect copy of the data originally transmitted, it passes that datagram to the application program. Thus, TCP abstracts the application’s communication from the underlying networking details.
TCP is used extensively by many of the Internet’s most popular applications, including the World Wide Web, E-mail, File Transfer Protocol, Secure, and some streaming media applications.
TCP is optimized for accurate delivery rather than timely delivery, and therefore, TCP sometimes incurs relatively long delays (in the order of seconds) while waiting for out-of-order messages or retransmissions of lost messages. It is not particularly suitable for real-time applications such as Voice over IP. For such applications, protocols like the Real-time Transport Protocol (RTP) running over the User Datagram Protocol(UDP) are usually recommended instead.
TCP is a reliable stream delivery service that guarantees delivery of a data stream sent from one host to another without duplication or losing data.
Since packet transfer is not reliable, a technique known as positive acknowledgment with retransmission is used to guarantee reliability of packet transfers. This fundamental technique requires the receiver to respond with an acknowledgment message as it receives the data. The sender keeps a record of each packet it sends, and waits for acknowledgment before sending the next packet. The sender also keeps a timer from when the packet was sent, and retransmits a packet if the timer expires. The timer is needed in case a packet gets lost or corrupted.
TCP consists of a set of rules: for the protocol, that are used with the Internet Protocol, and for the IP, to send data “in a form of message units” between computers over the Internet. At the same time that IP takes care of handling the actual delivery of the data, TCP takes care of keeping track of the individual units of data transmission, called segments, that a message is divided into for efficient routing through the network. For example, when an HTML file is sent from a Web server, the TCP software layer of that server divides the sequence of bytes of the file into segments and forwards them individually to the IP software layer (Internet Layer). The Internet Layer encapsulates each TCP segment into an IP packet by adding a header which includes (among other data) the destination IP address. Even though every packet has the same destination address, they can be routed on different paths through the network. When the client program on the destination computer receives them, the TCP layer (Transport Layer) reassembles the individual segments and ensures they are correctly ordered and error free as it streams them to an application.
Protocol Operation
TCP protocol operations may be divided into three phases. Connections must be properly established in a multi-step handshake process (connection establishment) before entering the data transfer phase. After data transmission is completed, the connection termination closes established virtual circuits and releases all allocated resources.
A TCP connection is managed by an operating system through a programming interface that represents the local end-point for communications, the Internet socket. During the lifetime of a TCP connection it undergoes a series of state changes:
1. LISTEN : In case of a server, waiting for a connection request from any remote client.
2. SYN-SENT : waiting for the remote peer to send back a TCP segment with the SYN and ACK flags set. (usually set by TCP clients)
3. SYN-RECEIVED : waiting for the remote peer to send back an acknowledgment after having sent back a connection acknowledgment to the remote peer. (usually set by TCP servers)
4. ESTABLISHED : the port is ready to receive/send data from/to the remote peer.
5. FIN-WAIT-1
6. FIN-WAIT-2
7. CLOSE-WAIT
8. CLOSING
9. LAST-ACK
10. TIME-WAIT : represents waiting for enough time to pass to be sure the remote peer received the acknowledgment of its connection termination request. According to RFC 793 a connection can stay in TIME-WAIT for a maximum of four minutes.
11. CLOSED
Describe the impact of applications (Voice Over IP and Video Over IP) on a network 1
Voice over Internet Protocol (VoIP) is a general term for a family of transmission technologies for delivery of voice communications over IP networks such as the Internet or other packet-switched networks. Other terms frequently encountered and synonymous with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone.
Internet telephony refers to communications services — voice, facsimile, and/or voice-messaging applications — that are transported via the Internet, rather than the public switched telephone network (PSTN). The basic steps involved in originating an Internet telephone call are conversion of the analogue voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet; the process is reversed at the receiving end.
VoIP systems employ session control protocols to control the set-up and tear-down of calls as well as audio codec’s which encode speech allowing transmission over an IP network as digital audio via an audio stream. Codec use is varied between different implementations of VoIP (and often a range of codec’s are used); some implementations rely on narrowband and compressed speech, while others support high fidelity stereo codec’s.
Challenges
Quality of service (QoS)
Because the underlying IP network is inherently less reliable, in contrast to the circuit-switched public telephone network, and does not provide a mechanism to ensure that data packets are delivered in sequential order, or provide quality-of-service (QoS) guarantees, VoIP implementations may face problems mitigating latency and jitter.
Voice, and all other data, travel in packets over IP networks with fixed maximum capacity. This system is more prone to congestion and DoS attacks than traditional switched systems; a circuit switched system of insufficient capacity will refuse new connections while carrying the remainder without impairment, while the quality of real-time data such as telephone conversations on packet-switched networks degrades dramatically.
Fixed delays cannot be controlled (as they are caused by the physical distance the packets travel), however some delays can be minimized by marking voice packets as being delay-sensitive (see, for example, DiffServ). Fixed delays are especially problematic when satellite circuits are involved, because of long round-trip propagation delay (400–600 milliseconds for links through geostationary satellites).
A cause of packet loss and delay is congestion, which can be avoided by means of teletraffic engineering.
The receiving node must restructure IP packets that may be out of order, delayed or missing, while ensuring that the audio stream maintains a proper time consistency. Variation in delay is called jitter. The effects of jitter can be mitigated by storing voice packets in a jitter buffer upon arrival and before producing analogue audio, although this further increases delay. This avoids a condition known as buffer under-run, in which the voice engine is missing audio since the next voice packet has not yet arrived. When IP packets are lost or delayed at any point in the network between VoIP users there will be a momentary dropout of voice if all packet delay and loss mechanisms cannot compensate.
It has been suggested to rely on the packetized nature of media in VoIP communications and transmit the stream of packets from the source phone to the destination phone simultaneously across different routes (multi-path routing). In such a way, temporary failures have less impact on the communication quality.
In capillary routing it has been suggested to use at the packet level Fountain codes or particularly raptor codes for transmitting extra redundant packets making the communication more reliable.
A number of protocols have been defined to support the reporting of QoS/QoE for VoIP calls. These include RTCP Extended Report(RFC 3611), SIP RTCP Summary Reports, H.460.9 Annex B (forH.323), H.248.30 and MGCP extensions. The RFC 3611 VoIP Metrics block is generated by an IP phone or gateway during a live call and contains information on packet loss rate, packet discard rate (because of jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end system delay, signal / noise / echo level, Mean Opinion Scores (MOS) and R factors and configuration information related to the jitter buffer.
RFC 3611 VoIP metrics reports are exchanged between IP endpoints on an occasional basis during a call, and an end of call message sent via SIP RTCP Summary Report or one of the other signalling protocol extensions. RFC 3611 VoIP metrics reports are intended to support real time feedback related to QoS problems, the exchange of information between the endpoints for improved call quality calculation and a variety of other applications.
Describe the impact of applications (Voice Over IP and Video Over IP) on a network 2
A number of protocols that deal with the data link layer and physical layer include quality-of-service mechanisms that can be used to ensure that applications like VoIP work well even in congested scenarios. Some examples include:
IEEE 802.11e is an approved amendment to the IEEE 802.11 standard that defines a set of quality-of-service enhancements for wireless LAN applications through modifications to the Media Access Control (MAC) layer. The standard is considered of critical importance for delay-sensitive applications, such as Voice over Wireless IP.
IEEE 802.1p defines 8 different classes of service (including one dedicated to voice) for traffic on layer-2 wired Ethernet.
The ITU-T G.hn standard, which provides a way to create a high-speed (up to 1 gigabit per second) Local area network using existing home wiring (power lines, phone lines and coaxial cables). G.hn provides QoS by means of “Contention-Free Transmission Opportunities” (CFTXOPs) which are allocated to flows (such as a VoIP call) which require QoS and which have negotiated a “contract” with the network controller.
Susceptibility to power failure
Telephones for traditional residential analogue service are usually connected directly to telephone company phone lines which provide direct current to power most basic analogue handsets independently of locally available power.
IP Phones and VoIP telephone adapters connect to routers or cable modems which typically depend on the availability of mains electricity or locally generated power. Some VoIP service providers use customer premise equipment (e.g., cable modems) with battery-backed power supplies to assure uninterrupted service for up to several hours in case of local power failures. Such battery-backed devices typically are designed for use with analogue handsets.
The susceptibility of phone service to power failures is a common problem even with traditional analogue service in areas where many customers purchase modern handset units that operate wirelessly to a base station, or that have other modern phone features, such as built-in voicemail or phone book features.
Emergency calls
The nature of IP makes it difficult to locate network users geographically. Emergency calls, therefore, cannot easily be routed to a nearby call centre. Sometimes, VoIP systems may route emergency calls to a non-emergency phone line at the intended department.
A fixed line phone has a direct relationship between a telephone number and a physical location. A telephone number represents one pair of wires that links a location to the telephone company’s exchange. Once a line is connected, the telephone company stores the home address that relates to the wires, and this relationship will rarely change. If an emergency call comes from that number, then the physical location is known.
In the IP world, it is not so simple. A broadband provider may know the location where the wires terminate, but this does not necessarily allow the mapping of an IP address to that location. IP addresses are often dynamically assigned, so the ISP may allocate an address for online access, or at the time a broadband router is engaged. The ISP recognizes individual IP addresses, but does not necessarily know what physical location to which it corresponds. The broadband service provider knows the physical location, but is not necessarily tracking the IP addresses in use.
There are more complications, since IP allows a great deal of mobility. For example, a broadband connection can be used to dial a virtual private network that is employer-owned. When this is done, the IP address being used will belong to the range of the employer, rather than the address of the ISP, so this could be many kilometres away or even in another country. To provide another example: if mobile data is used, e.g., a 3G mobile handset or USB wireless broadband adapter, then the IP address has no relationship with any physical location, since a mobile user could be anywhere that there is network coverage, even roaming via another mobile phone company.
In short, there is no relationship between IP address and physical location, so the address itself reveals no useful information for the emergency services.
At the VoIP level, a phone or gateway may identify itself with a SIP registrar by using a username and password. So in this case, the Internet Telephony Service Provider (ITSP) knows that a particular user is online, and can relate a specific telephone number to the user. However, it does not recognize how that IP traffic was engaged. Since the IP address itself does not necessarily provide location information presently, today a “best efforts” approach is to use an available database to find that user and the physical address the user chose to associate with that telephone number—clearly an imperfect solution.
VoIP Enhanced 911 (E911) is another method by which VoIP providers in the United States are able to support emergency services. The VoIP E911 emergency-calling system associates a physical address with the calling party’s telephone number as required by the Wireless Communications and Public Safety Act of 1999. All “interconnected” VoIP providers (those that provide access to the PSTN system) are required to have E911 available to their customers. VoIP E911 service generally adds an additional monthly fee to the subscriber’s service per line, similar to analogue phone service. Participation in E911 is not required and customers can opt-out or disable E911 service on their VoIP lines, if desired. VoIP E911 has been successfully used by many VoIP providers to provide physical address information to emergency service operators.
One shortcoming of VoIP E911 is that the emergency system is based on a static table lookup. Unlike in mobile phones, where the location of an E911 call can be traced using Assisted GPS or other methods, the VoIP E911 information is only accurate so long as subscribers are diligent in keeping their emergency address information up-to-date.
Describe the impact of applications (Voice Over IP and Video Over IP) on a network 3
With the current separation of the Internet and the PSTN, a certain amount of redundancy is provided. An Internet outage does not necessarily mean that a voice communication outage will occur simultaneously, allowing individuals to call for emergency services and many businesses to continue to operate normally. In situations where telephone services become completely reliant on the Internet infrastructure, a single-point failure can isolate communities from all communication, including Enhanced 911 and equivalent services in other locales.
PSTN integration
E.164 is a global numbering standard for both the PSTN and PLMN. Most VoIP implementations support E.164 to allow calls to be routed to and from VoIP subscribers and the PSTN/PLMN. VoIP implementations can also allow other identification techniques to be used. For example, Skype allows subscribers to choose “Skype names” (usernames) whereas SIP implementations can use URIs similar to email addresses. Often VoIP implementations employ methods of translating non-E.164 identifiers to E.164 numbers and vice-versa, such as the Skype-In service provided by Skype and the ENUM service in IMS and SIP.
Echo can also be an issue for PSTN integration. Common causes of echo include impedance mismatches in analogue circuitry and acoustic coupling of the transmit and receive signal at the receiving end.
Security
Voice over Internet Protocol telephone systems (VoIP) are susceptible to attacks as are any internet-connected devices. This means that hackers who know about these vulnerabilities can institute denial-of-service attacks, harvest customer data, record conversations and break into voice mailboxes.
Another challenge is routing VoIP traffic through firewalls and network address translators. Private Session Border Controllers are used along with firewalls to enable VoIP calls to and from protected networks. Skype uses a proprietary protocol to route calls through other Skype peers on the network, allowing it to traverse symmetric NATs and firewalls. Other methods to traverse NATs involve using protocols such as STUN or ICE.
Many consumer VoIP solutions do not support encryption, although having a secure phone is much easier to implement with VoIP than traditional phone lines. As a result, it is relatively easy to eavesdrop on VoIP calls and even change their content. An attacker with a packet sniffer could intercept your VoIP calls if you are not on a secure VLAN.
There are open source solutions, such as Wireshark, that facilitate sniffing of VoIP conversations. A modicum of security is afforded by patented audio codec’s in proprietary implementations that are not easily available for open source applications, however such security through obscurity has not proven effective in other fields. Some vendors also use compression to make eavesdropping more difficult. However, real security requires encryption and cryptographic authentication which are not widely supported at a consumer level. The existing security standard Secure Real-time Transport Protocol (SRTP) and the new ZRTP protocol are available on Analogue Telephone Adapters(ATAs) as well as various soft-phones. It is possible to use IPsec to secure P2P VoIP by using opportunistic encryption. Skype does not use SRTP, but uses encryption which is transparent to the Skype provider. In 2005, Skype invited a researcher, Dr Tom Berson, to assess the security of the Skype software, and his conclusions are available in a published report.
The Voice VPN solution provides secure voice for enterprise VoIP networks by applying IPSec encryption to the digitized voice stream.
Securing VoIP
To prevent the above security concerns the government and military organizations are using; Voice over Secure IP (VoSIP), Secure Voice over IP (SVoIP), and Secure Voice over Secure IP (SVoSIP) to protect confidential, and/or classified VoIP communications. Secure Voice over IP is accomplished by encrypting VoIP with Type 1 encryption. Secure Voice over Secure IP is accomplished by using Type 1 encryption on a classified network, like SIPRNet. Public Secure VoIP is also available with free GNU programs.
Caller ID
Caller ID support among VoIP providers varies, although the majority of VoIP providers now offer full Caller ID with name on outgoing calls.
In a few cases, VoIP providers may allow a caller to spoof the Caller ID information, potentially making calls appear as though they are from a number that does not belong to the caller Business grade VoIP equipment and software often makes it easy to modify caller ID information. Although this can provide many businesses great flexibility, it is also open to abuse.
Compatibility with traditional analogue telephone sets
Some analogue telephone adapters do not decode pulse dialling from older phones. The VoIP user may use a pulse-to-tone converter, if needed.
Fax handling
Support for sending faxes over VoIP implementations is still limited. The existing voice codec’s are not designed for fax transmission; they are designed to digitize an analogue representation of a human voice efficiently. However, the inefficiency of digitizing an analogue representation (modem signal) of a digital representation (a document image) of analogue data (an original document) more than negates any bandwidth advantage of VoIP. In other words, the fax “sounds” simply don’t fit in the VoIP channel. An alternative IP-based solution for delivering fax-over-IP called T.38 is available.
The T.38 protocol is designed to work like a traditional fax machine and can work using several configurations. The fax machine could be a traditional fax machine connected to the PSTN, or an ATA box (or similar). It could be a fax machine with an RJ-45 connector plugged straight into an IP network, or it could be a computer pretending to be a fax machine. Originally, T.38 was designed to use UDP and TCP transmission methods across an IP network. The main difference between using UDP and TCP methods for a FAX is the real time streaming attributes. TCP is better suited for use between two IP devices. However, older fax machines, connected to an analogue system, benefit from UDP near real-time characteristics.
There have been updated versions of T.30 to resolve the fax over IP issues, which is the core fax protocol. Some new fax machines have T.38 built-in capabilities which allow the user to plug right into the network with minimal configuration changes. A unique feature of T.38 is that each packet contains a copy of the main data in the previous packet. This is an option and most implementations seem to support it. This forward error correction scheme makes T.38 far more tolerant of dropped packets than VoIP. With T.38, two successive lost packets are needed to actually lose any data. The data you lose will only be a small piece, but with the right settings and error correction mode, there is a high probability that you will receive the whole transmission.
Tweaking the settings on the T.30 and T.38 protocols could also turn your unreliable fax into a robust machine. Some fax machines pause at the end of a line to allow the paper feed to catch up. This is good news for packets that were lost or delayed because it gives them a chance to catch up. However, were this to happen on every line, your fax transmittal would take a long time. The end system can completely buffer the incoming fax data before displaying or printing the fax image.
Support for other telephony devices
Another challenge for VoIP implementations is the proper handling of outgoing calls from other telephony devices such as DVR boxes, satellite television receivers, alarm systems,
conventional modems and other similar devices that depend on access to a PSTN telephone line for some or all of their functionality.
These types of calls sometimes complete without any problems, but in other cases they fail. If VoIP and mobile substitution becomes very popular, some ancillary equipment makers may be forced to redesign equipment, because it would no longer be possible to assume a conventional PSTN telephone line would be available in consumer’s homes.
Video Over IP
Professional video over IP systems use some existing standard video codec to reduce the program material to a bitstream (e.g., an MPEG transport stream), and then to use an Internet Protocol(IP) network to carry that bitstream encapsulated in a stream of IP packets. This is typically accomplished using some variant of the RTP protocol.
Carrying professional video over IP networks has special challenges compared to most non-time-critical IP traffic. Many of these problems are similar to those encountered in voice over IP, but to a much higher level of engineering requirements. In particular, there are very strict quality of service (QoS) requirements which must be fulfilled for use in professional broadcast environments.
Packet Loss
Since even well-engineered IP networks tend to have a small residual packet loss rate caused by low-probability statistical congestion events and amplification of bit errors in the underlying hardware, most professional solutions use some kind of forward error correction to ensure that the encoded video stream can be reconstructed even if a few packets are lost. This is typically applied at the packet level, since the encapsulated video bitstream is typically only designed to tolerate low levels of bit or burst errors, rather than the loss of whole packets. Resending packets is not an option because of the sequential nature of the underlying video signal. For live video, a resent packet would arrive well after the arrival of the next frame of video.
Network Delay Variation
Network delay variation can be kept to a minimum by using a high-speed network backbone, and ensuring that video traffic does not encounter excessive queue delays. This is typically done by either ensuring that the network is not too close to its full capacity, or that video traffic is prioritized using traffic engineering techniques.
The remaining delay variation can be removed by buffering, at the expense of added time delay. If forward error correction is used, a small proportion of packets arriving after the deadline can be tolerated, since they can be dealt with by being discarded on receipt, and then treated in the same way as lost packets. Added time delay is particularly unwelcome in PTZ cameras as it makes operator control difficult at values over 250ms.
Timing Reconstruction
The other problem presented by latency variation is that it makes synchronization more complex by making the recovery of the underlying timing of the video signal far more difficult. This is typically solved by genlocking both ends of the system to external station sync signals, typically generated from sources such as GPS or atomic clocks, thus only requiring the extraction of coarse timing information at the receiving end in order to achieve high-quality video synchronization. The extraction of coarse timing data is typically done using a phase locked loop with a long time constant.
Adequate Bandwidth
Even with packet loss mitigation, video over IP will only work if the network is capable of carrying the content with some reasonable maximum packet loss rate. In practice, this means that video over IP will not work on overloaded networks. Since IP does not of itself offer any traffic guarantees, this must be applied at the network engineering level. One approach to this is the “quantity of service” approach which simply allocates sufficient bandwidth to video-carrying traffic that it will not congest under any possible load pattern. Other approaches include dynamic reduction in frame rate or resolution, network admission control, bandwidth reservation, traffic shaping, and traffic prioritization techniques, which require more complex network engineering, but will work when the simple approach of building a non-blocking network is not possible.
Use by Security Industry
Within the security products industry, IP-based Closed Circuit Television (CCTV) is making gains on the analogue market. Key components of IP-based CCTV remain consistent with analogue technologies: image capture, with a combination of IP-based cameras or analogue cameras using IP-based encoders; image transmission; Storage and Retrieval, which uses technologies such as RAID arrays and iSCSI for recorded and indexed video; and video management, which affords web browser-enabled management and control of IP-based CCTV systems.
One key advantage of IP-based CCTV is the ability to use network infrastructure, providing adequate bandwidth and availability of switching and routing, rather than coaxial cabling. However, running bandwidth-intensive surveillance video over corporate data networks is a point of organizational contention, depending on the potential impact on network performance.
A class of companies (including Aimetis, Milestone Systems, videoNEXT, Verint and others) produce Video Management Software to help manage capture and storage of video content. Digital video also makes possible Video Content Analysis, which allows automatic detection and identification of various kinds of objects or motion. Companies in this market include Aimetis, VideoIQ, ObjectVideo, and Cernium.
Also another emerging model is off-site storage of video surveillance video. These online surveillance providers are utilizing cloud computing based technologies to consolidate multi-site surveillance video over the web. ByRemote, Iveda Solutions and Secure-I provide these services from an off-site data centre.
Several manufacturers of CCTV equipment, such as Dallmeier, Axis Communications, General Electric, Bosch, Pelco, Siemens and Sanyo are steadily integrating IP network technology into their product portfolios.
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