Intranet

An intranet is a private computer network that uses Internet Protocol technologies to securely share any part of an organization's information or network operating system within that organization. The term is used in contrast to internet, a network between organizations, and instead refers to a network within an organization. Sometimes the term refers only to the organization's internal website, but may be a more extensive part of the organization's information technology infrastructure.

DXX Technology
Digital Cross Connect or DXX is a system that covert the signals from one communication medium to another. Typically DXX was used to convert the high level TDM signals in telephone systems for example DSI bit streams into another DSI streams. DXX devices are available for both T carrier and E- Carrier based circuits. DXX are also used to connect two different data streams like Frame Relay and Serial Communication. DXX circuits can be used switch traffic from one circuit to another in case of a network failure among the network. For example if a frame relay network gets deactivated for any reason, DXX can convert the traffic to another network like ATM. In this case DXX switches can manipulate the traffic circuits between different networks and network types. It is also used to provide interoperability between network devices and network speeds. With DXX, higher levels of flexibility can be provided which in any other case can cause greater costs to the network operations. This flexibility comes with DXX at lower costs. With DXX, synchronization becomes possible with higher network seeds and lower network speeds. One important thing to be noted is that DXX are not packet

network switches. DXX switch between circuits and are a special means for circuit switched networks. This arrangement is also inevitable in longer networks like the ones that expand over weeks or months. This type of arrangement cant be made in packet networks that operate over short period of time. DXX are different from packet switches in a way that packet switches provide the transportation to different network destinations while in the case of DXX, it has a specific set of destinations to which it switches traffic in circuits. Digital Cross connects can connect extremely high speed networks with lower speed networks to achieve a mutually synchronized speed for common data transformation in the network. SXX are also capable of interconnect fiber optic equipment and D1/ T1/ E1 etc. It can even provide compatibility between rather new SDH, PDH and SONET. If the speed of SDH is 40 mbps and the speed id T1 is 2 Mbps then with the help of two DXXs, Two T1as and one SDH, you can divert two different streams of T1 to SDH and vice versa making a communication of 4 Mbps possible through the network.

What is E1 line
In telecommunications, where a single physical wire can be used to carry many simultaneous voice conversations, worldwide standards have been created and deployed. E-carrier system, which is revised and improved version of the earlier American T-carrier technology. Now it is widely used in almost all countries outside USA, Canada and Japan. The line data rate for E1 is 2.048 Mbit/s (full duplex) which is split into 32 time slots, each being allocated 8 bits in turn. It is a ideal for voice traffic because voice is sampled at the same 8khz rate so E1 line can carry 32 simultaneous voice conversions.

What is E3 line
Businesses that connect to the Internet using dedicated access services are experiencing traffic growth due to increased interest and dependence on resourceintensive applications. To improve availability of network-based applications, network architects and designers are searching for solutions that can extend their current bandwidth and transmission capacity. In response to higher bandwidth and capacity demand, carriers and ISPs are now offering E3 (34 Mbps) dedicated access connections. Anticipating these trends, Cisco Systems offers the Cisco 12000 Series 6-Port and 12-Port E3 Line Cards; designed to simplify the deployment and delivery of E3 leasedline services with the Cisco 12000 series Internet router.

Digital Subscriber Line Access Multiplexer

A Digital Subscriber Line Access Multiplexer (DSLAM, often pronounced deeslam) allows telephone lines to make faster connections to the Internet. It is a network device, located in the telephone exchanges of the service providers, that connects multiple customer Digital Subscriber Lines (DSLs) to a high-speed Internet backbone line using multiplexing techniques. By placing remote DSLAMs at locations remote to the telephone exchange, telephone companies provide DSL service to locations previously beyond effective range.

Role of the DSLAM

The DSLAM equipment at the telephone company (telco) collects the data from its many modem ports and aggregates their voice and data traffic into one complex composite "signal" via multiplexing. Depending on its device architecture and setup, a DSLAM aggregates the DSL lines over its Asynchronous Transfer Mode (ATM), frame relay, and/or Internet Protocol network (i.e., an IP-DSLAM using PTM-TC [Packet Transfer Mode - Transmission Convergence]) protocol(s) stack. The aggregated traffic is then directed to a telco's backbone switch, via an access network (AN) also called a Network Service Provider (NSP) at up to 10 Gbit/s data rates.The DSLAM acts like a network switch since its functionality is at Layer 2 of the OSI model. Therefore it cannot re-route traffic between multiple IP networks, only between ISP devices and end-user connection points. The DSLAM traffic is switched to a Broadband Remote Access Server where the end user traffic is then routed across the ISP network to the Internet. Customer Premises Equipment that interfaces well with the DSLAM to which it is connected may take advantage of enhanced telephone voice and data line signaling features and the bandwidth monitoring and compensation capabilities it supports. A DSLAM may or may not be located in the telephone company's central office, and may also serve multiple data and voice customers within a neighborhood Serving Area Interface (SAI), sometimes in conjunction with a digital loop carrier. DSLAMs are also used by hotels, lodges, residential neighborhoods, and other businesses operating their own private telephone exchange. In addition to being a data switch and multiplexer, a DSLAM is also a large collection of modems. Each modem on the aggregation card communicates with a single subscriber's DSL modem. This modem functionality is integrated into the DSLAM itself instead of being done via an external device like a traditional computer modem. Like traditional voice-band modems, a DSLAM's integrated DSL modems usually have the ability to probe the line and to adjust themselves to electronically or digitally compensate for forward echoes and other bandwidth-limiting factors in order to move data at the maximum connection rate capability of the subscriber's physical line. This compensation capability also takes advantage of the better performance of "balanced line" DSL connections, providing capabilities for LAN segments longer than physically-similar unshielded twisted pair (UTP) Ethernet connections, since the balanced line type is generally required for its hardware to function correctly. This is due to the nominal line impedance (measured in Ohms but comprising

both resistance and inductance) of balanced lines being somewhat lower than that of UTP, thus supporting 'weaker' signals (however the solid-state electronics required to construct such digital interfaces is more costly).

Leased line
A leased line is service contract between a provider and a customer, whereby the provider agrees to deliver a symmetric telecommunications line connecting two locations in exchange for a monthly rent (hence the term lease). It is sometimes known as a 'Private Circuit' or 'Data Line' in the UK or as CDN (Circuito Diretto Numerico) in Italy. Unlike traditional PSTN lines it does not have a telephone number, each side of the line being permanently connected to the other. Leased lines can be used for telephone, data or Internet services. Some are ringdown services, and some connect two PBXes. A permanent telephone connection between two points set up by a telecommunications common carrier. Typically, leased lines are used by businesses to connect geographically distant offices. Unlike dial-up connections, a leased line is always active. The fee for the connection is a fixed monthly rate. The primary factors affecting the monthly fee are distance between end points and the speed of the circuit. Because the connection doesn't carry anybody else's communications, the carrier can assure a given level of quality.

Virtual Connection
In telecommunications and computer networks, a virtual circuit (VC),

synonymous with virtual connection and virtual channel, is a connection oriented communication service that is delivered by means of packet mode communication. After a connection or virtual circuit is established between two nodes or application processes, a bit stream or byte stream may be delivered between the nodes. A virtual circuit protocol hides the division into segments, packets or frames from higher level protocols.

Transmission Control Protocol (TCP), where a reliable virtual circuit is established on top of the underlying unreliable and connectionless IP protocol. The virtual circuit is identified by the source and destination network socket address pair, i.e. the sender and receiver IP address and port number. Guaranteed QoS is not provided. SCTP, where a virtual circuit is established on top of either the IP protocol or the UDP protocol.

Examples of network layer and datalink layer virtual circuit protocols, where data always is delivered over the same path:

X.25, where the VC is identified by a virtual channel identifier (VCI). X.25

provides reliable node-to-node communication and guaranteed QoS. Frame relay, where the VC is identified by a VCI. Frame relay is unreliable, but may provide guaranteed QoS. Asynchronous Transfer Mode (ATM), where the circuit is identified by a virtual path identifier (VPI) and virtual channel identifier (VCI) pair. ATM is unreliable, but may provide guaranteed QoS.

ATM
Asynchronous Transfer Mode is a cell-based switching technique that uses asynchronous time division multiplexing. It encodes data into small fixed-sized cells (cell relay) and provides data link layer services that run over OSI Layer 1 physical links. This differs from other technologies based on packet-switched networks (such as the Internet Protocol or Ethernet), in which variable sized packets (known as frames when referencing Layer 2) are used. ATM exposes properties from both circuit switched and small packet switched networking, making it suitable for wide area data networking as well as real-time media transport. ATM uses a connection-oriented model and establishes a virtual circuit between two endpoints before the actual data exchange begins. At the time of the design of ATM, 155 Mbit/s SDH (135 Mbit/s payload) was considered a fast optical network link, and many PDH links in the digital network were considerably slower, ranging from 1.544 to 45 Mbit/s in the USA, and 2 to 34 Mbit/s in Europe. At this rate, a typical full-length 1500 byte (12000-bit) data packet would take 77.42 s to transmit. In a lower-speed link, such as a 1.544 Mbit/s T1 link; a 1500 byte packet would take up to 7.8 milliseconds. A queuing delay induced by several such data packets might exceed the figure of 7.8 ms several times over, in addition to any packet generation delay in the shorter speech packet. This was clearly unacceptable for speech traffic, which needs to have low jitter in the data stream being fed into the codec if it is to produce good-quality sound.

Network switch
A network switch or switching hub is a computer networking device that connect network segments. The term commonly refers to a network bridge that processes and routes data at the data link layer (layer 2) of the OSI model. Switches that additionally process data at the network layer (layer 3 and above) are often referred to as Layer 3 switches or multilayer switches. The term network switch does not generally encompass unintelligent or passive network devices such as hubs and repeaters. The first Ethernet switch was introduced by Kalpana in 1990.

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 a 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 Linksys 8-port and 48-port devices. User devices may also include a telephone interface for VoIP. A standard 10/100 Ethernet switch operates at the data-link layer of the OSI model to create a different collision domain for each switch port. If you have 4 computers (e.g., A, B, C, and D) on 4 switch ports, then A and B can transfer data back and forth, while C and D also do so simultaneously, and the two "conversations" will not interfere with one another. In the case of a "hub," they would all share the bandwidth and run in Half duplex, resulting in collisions, which would then necessitate retransmissions. Using a switch is called microsegmentation. This allows you to have dedicated bandwidth on point-to-point connections with every computer and to therefore run in Full duplex 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. A network bridge, operating at the Media Access Control (MAC) sublayer 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 reconverge. 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 linkstate routing technology to the layer-2 bridging problem. Devices which implement TRILL, called RBridges, 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 performanceincreasing methods when used on "switch" products with the same input and output port speeds: 1. Store and forward: The switch buffers and, typically, performs a checksum on each frame before forwarding it. 2. 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. 3. 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. 4. 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.Cypress Semiconductor design and manufacturing company along with TPACK offers the flexibility to cope with various system architecture for Ethernet switches through reference design. The reference design involves TPX4004 and CY7C15632KV18 72-Mbit SRAMs.

Address Resolution Protocol

The Address Resolution Protocol (ARP) is a computer networking protocol for determining a network host's link layer or hardware address when only its Internet Layer (IP) or Network Layer address is known. This function is critical in local area networking as well as for routing internetworking traffic across gateways (routers) based on IP addresses when the next-hop router must be determined. ARP was defined by RFC 826 in 1982. It is Internet Standard STD 37. ARP has been implemented in many types of networks, such as Internet Protocol (IP) network, CHAOS, DECNET, Xerox PARC Universal Packet, Token Ring, FDDI, IEEE 802.11 and other LAN technologies, as well as the modern high capacity networks, such as Asynchronous Transfer Mode (ATM). Due to the overwhelming prevalence of IPv4 and Ethernet in general networking, ARP is most frequently used to translate IPv4 addresses (OSI Layer 3) into Ethernet MAC addresses (OSI Layer 2).

Ethernet
Ethernet is a family of frame-based computer networking technologies for local area networks (LANs). The name came from the physical concept of the ether. It defines a number of wiring and signaling standards for the Physical Layer of the OSI networking model as well as a common addressing format and Media Access Control at the Data Link Layer. Ethernet is standardized as IEEE 802.3. The combination of the twisted pair versions of Ethernet for connecting end systems to the network, along with the fiber optic versions for site backbones, is the most widespread wired LAN technology. It has been used from around 1980 to the present, largely replacing competing LAN standards such as token ring, FDDI, and ARCNET.

Crossover cable
A crossover cable connects two devices of the same type, for example DTE-DTE or DCE-DCE, usually connected asymmetrically (DTE-DCE), by a modified cable called a crosslink. Such distinction of devices was introduced by IBM. The crossing wires in a cable or in a connector adaptor allows:

connecting two devices directly, output of one to input of the other, letting two terminal (DTE) devices communicate without an interconnecting hub knot, i.e. PCs, linking two or more hubs, switches or router (DCE) together, possibly to work as one wider device.

Examples

a Null modem of RS-232 Ethernet crossover cable

Rollover cable A loopback is a type of degraded "one side crosslinked connection" connecting a port to itself, usually for test purposes.

Use crossover cables for the following connections: Switch to switch, Switch to hub, Hub to hub, Router to router, PC to PC, Router to PC

Frame Relay
Frame Relay is a standardized wide area networking technology that specifies the physical and logical link layers of digital telecommunications channels using a packet switching methodology. Originally designed for transport across Integrated Services Digital Network (ISDN) infrastructure, it may be used today in the context of many other network interfaces. Network providers commonly implement Frame Relay for voice (VoFR) and data as an encapsulation technique, used between local area networks (LANs) over a wide area network (WAN). Each end-user gets a private line (or leased line) to a frame-relay node. The frame-relay network handles the transmission over a frequently-changing path transparent to all end-users. With the advent of MPLS, VPN and dedicated broadband services such as cable modem and DSL, the end may loom for the Frame Relay protocol and encapsulation. However many rural areas remain lacking DSL and cable modem services. In such cases the least expensive type of "always-on" connection remains a 64-kbit/s frame-relay line. Thus a retail chain, for instance, may use Frame Relay for connecting rural stores into their corporate WAN. The designers of Frame Relay aimed to a telecommunication service for cost-efficient data transmission for intermittent traffic between local area networks (LANs) and between end-points in a wide area network (WAN). Frame Relay puts data in variable-size units called "frames" and leaves any necessary error-correction (such as re-transmission of data) up to the end-points. This speeds up overall data transmission. For most services, the network provides a permanent virtual circuit (PVC), which means that the customer sees a continuous, dedicated connection without having to pay for a full-time leased line, while the service-provider figures out the route each frame travels to its destination and can charge based on usage. Frame Relay has its technical base in the older X.25 packet-switching technology, designed for transmitting data on analog voice lines. Unlike X.25, whose designers expected analog signals, Frame Relay offers a fast packet technology, which means that the protocol does not attempt to correct errors. When a Frame Relay network detects an error in a frame, it simply drops that frame. It requires a dedicated connection during the transmission period. Frame Relay does not provide an ideal path for voice or video transmission, both of which require a steady flow of transmissions. However, under certain circumstances, voice and video transmission do use Frame Relay. Frame Relay relays packets at the data link layer (layer 2) of the Open Systems Interconnection (OSI) model rather than at the network layer (layer 3). A frame can incorporate packets from different protocols such as Ethernet and X.25. It varies in size up to a thousand bytes or more.

Frame Relay originated as an extension of Integrated Services Digital Network (ISDN). Its designers aimed to enable a packet-switched network to transport the circuit-switched technology. The technology has become a stand-alone and cost-effective means of creating a WAN. Frame Relay switches create virtual circuits to connect remote LANs to a WAN. The Frame Relay network exists between a LAN border device, usually a router, and the carrier switch. The technology used by the carrier to transport the data between the switches is variable and changes between carrier (i.e. Frame Relay does not rely directly on the transportation mechanism to function). The sophistication of the technology requires a thorough understanding of the terms used to describe how Frame Relay works. Without a firm understanding of Frame Relay, it is difficult to troubleshoot its performance. The Frame Relay network uses a simplified protocol at each switching node. It achieves simplicity by omitting link-by-link flow-control. As a result, the offered load has largely determined the performance of Frame Relay networks. When offered load is high, due to the bursts in some services, temporary overload at some Frame Relay nodes causes a collapse in network throughput. Therefore, frame-relay networks require some effective mechanisms to control the congestion. Congestion control in frame-relay networks includes the following elements: 1. Admission Control. This provides the principal mechanism used in Frame Relay to ensure the guarantee of resource requirement once accepted. It also serves generally to achieve high network performance. The network decides whether to accept a new connection request, based on the relation of the requested traffic descriptor and the network's residual capacity. The traffic descriptor consists of a set of parameters communicated to the switching nodes at call set-up time or at service-subscription time, and which characterizes the connection's statistical properties. The traffic descriptor consists of three elements: 2. Committed Information Rate (CIR). The average rate (in bit/s) at which the network guarantees to transfer information units over a measurement interval T. This T interval is defined as: T = Bc/CIR. 3. Committed Burst Size (BC). The maximum number of information units transmittable during the interval T. 4. Excess Burst Size (BE). The maximum number of uncommitted information units (in bits) that the network will attempt to carry during the interval. Once the network has established a connection, the edge node of the Frame Relay network must monitor the connection's traffic flow to ensure that the actual usage of network resources does not exceed this specification. Frame Relay defines some restrictions on the user's information rate. It allows the network to enforce the end user's information rate and discard information when the subscribed access rate is exceeded.

Virtual LAN
A virtual LAN, commonly known as a VLAN, is a group of hosts with a common set of requirements that communicate as if they were attached to the same broadcast domain, regardless of their physical location. A VLAN has the same attributes as a physical LAN, but it allows for end stations to be grouped together even if they are not located on the same network switch. Network reconfiguration can be done through software instead of physically relocating devices. To physically replicate the functions of a VLAN, it would be necessary to install a separate, parallel collection of network cables and switches/hubs which are kept separate from the primary network. However unlike a physically separate network, VLANs must share bandwidth; two separate one-gigabit VLANs using a single one-gigabit interconnection can both suffer reduced throughput and congestion. VLANs are created to provide the segmentation services traditionally provided by routers in LAN configurations. VLANs address issues such as scalability, security, and network management. Routers in VLAN topologies provide broadcast filtering, security, address summarization, and traffic flow management. By definition, switches may not bridge IP traffic between VLANs as it would violate the integrity of the VLAN broadcast domain.

Understanding IP Addresses
An IP address is an address used to uniquely identify a device on an IP network. The address is made up of 32 binary bits which can be divisible into a network portion and host portion with the help of a subnet mask. The 32 binary bits are broken into four octets (1 octet = 8 bits). Each octet is converted to decimal and separated by a period (dot). For this reason, an IP address is said to be expressed in dotted decimal format (for example172.16.81.100). The value in each octet ranges from 0 to 255 decimal, or 00000000 11111111 binary. Here is how binary octets convert to decimal: The right most bit, or least significant bit, of an octet holds avalue of 20. The bit just to the left of that holds a value of 21. This continues until the leftmost bit, or most significant bit, which holds a value of 27. So if all binary bits are a one, the decimal equivalent would be 255 as shown here: 11111111 128 64 32 16 8 4 2 1 (128+64+32+16+8+4+2+1=255) And this is sample shows an IP address represented in both binary and decimal. 10. 1. 23. 19 (decimal) 00001010.00000001.00010111.00010011 (binary)

Class Diagram

Network Masks
A network mask helps you know which portion of the address identifies the network and which portion of the address identifies the node. Class A, B, and C networks have default masks, also known as natural masks, as shown here: Class A: 255.0.0.0 Class B: 255.255.0.0 Class C: 255.255.255.0

Class A Networks
Each Class A network address has an 8-bit network prefix, with the highest order bit set to 0 (zero) and a 7-bit network number, followed by a 24-bit host number. Today, Class A networks are referred to as /8s (pronounced slash eight or just eights) since they have an 8- bit network prefix. A maximum of 126 (27 -2) /8 networks can be defined. The calculation subtracts two because the /8 network 0.0.0.0 is reserved for use as the default route and the /8 network 127.0.0.0 (also written 127/8 or 127.0.0.0/8) is reserved for the loopback function. Each /8 supports a maximum of 224 -2 (16,777,214) hosts per network. The host calculation subtracts two because the all-0s (all zeros or this network) and all-1s (all ones or broadcast) host numbers may not be assigned to individual hosts. Since the /8 address block contains (2,147,483,648) individual addresses and the IPv4 address space contains a maximum of 232 (4,294,967,296) addresses, the /8 address space is 50 percent of the total IPv4 unicast address space.

Class B Networks
Each Class B network address has a 16-bit network prefix, with the two highest order bits set to 1-0 and a 14-bit network number, followed by a 16-bit host number. Class B networks are now referred to as /16s since they have a 16-bit network prefix. A maximum of 16,384 networks can be defined with up to 65,534 (216-2) hosts per network. Since the entire /16 address block contains 230 (1,073,741,824) addresses, it represents 25 percent of the total IPv4 unicast address space.

Class C Networks
Each Class C network address has a 24-bit network prefix, with the three highest order bits set to 1-1-0 and a 21-bit network number, followed by an 8-bit host number. Class C networks are now referred to as /24s since they have a 24-bit network prefix. A maximum of 2,097,152 networks can be defined with up to 254 (28-2) hosts per network.

Since the entire /24 address block contains 229 (536,870,912) addresses, it represents 12.5 percent (or oneeighth) of the total IPv4 unicast address space.

Network Mask of Class A

An IP address on a Class A network that has not been subnetted would have an address/mask pair similar to: 8.20.15.1 255.0.0.0. To see how the mask helps you identify the network and node parts of the address, convert the address and mask to binary numbers. 8.20.15.1 = 00001000.00010100.00001111.00000001 255.0.0.0 = 11111111.00000000.00000000.00000000 Once you have the address and the mask represented in binary, then identifying the network and host ID is easier. Any address bits which have corresponding mask bits set to 1 represent the network ID. Any address bits that have corresponding mask bits set to 0 represent the node ID. 8.20.15.1 = 00001000.00010100.00001111.00000001 255.0.0.0 = 11111111.00000000.00000000.00000000 net id | host id netid = 00001000 = 8 hostid = 00010100.00001111.00000001 = 20.15.1

Understanding Subnetting
Subnetting allows you to create multiple logical networks that exist within a single Class A, B, or C network. If you do not subnet, you are only able to use one network from your Class A, B, or C network, which is unrealistic. Each data link on a network must have a unique network ID, with every node on that link being a member of the same network. If you break a major network (Class A, B, or C) into smaller subnetworks, it allows you to create a network of interconnecting subnetworks. Each data link on this network would then have a unique network/subnetwork ID. Any device, or gateway, connecting n networks/subnetworks has n distinct IP addresses, one for each network / subnetwork that it interconnects. Each data link on a network must have a unique network ID, with every node on that link being a member of the same network. If you break a major network (Class A, B, or C) into smaller subnetworks, it allows you to create a network of interconnecting subnetworks. Each data link on this network would then have a unique network/subnetwork ID. Any device, or gateway, connecting n networks/subnetworks has n distinct IP addresses, one for each network / subnetwork that it interconnects. In order to subnet a network, extend the natural mask using some of the bits from the host ID portion of the address to create a subnetwork ID. For example, given a Class C

network of 204.17.5.0 which has a natural mask of 255.255.255.0, you can create subnets in this manner: 204.17.5.0 11001100.00010001.00000101.00000000 255.255.255.224 11111111.11111111.11111111.11100000 |sub| By extending the mask to be 255.255.255.224, you have taken three bits (indicated by "sub") from the original host portion of the address and used them to make subnets. With these three bits, it is possible to create eight subnets. With the remaining five host ID bits, each subnet can have up to 32 host addresses, 30 of which can actually be assigned to a device since host ids of all zeros or all ones are not allowed (it is very important to remember this). So, with this in mind, these subnets have been created. 204.17.5.0 255.255.255.224 host address range 1 to 30 204.17.5.32 255.255.255.224 host address range 33 to 62 204.17.5.64 255.255.255.224 host address range 65 to 94 204.17.5.96 255.255.255.224 host address range 97 to 126 204.17.5.128 255.255.255.224 host address range 129 to 158 204.17.5.160 255.255.255.224 host address range 161 to 190 204.17.5.192 255.255.255.224 host address range 193 to 222 204.17.5.224 255.255.255.224 host address range 225 to 254

CIDR
Classless Interdomain Routing (CIDR) was introduced to improve both address space utilization and routing scalability in the Internet. It was needed because of the rapid growth of the Internet and growth of the IP routing tables held in the Internet routers. CIDR moves way from the traditional IP classes (Class A, Class B, Class C, and so on). In CIDR, an IP network is represented by a prefix, which is an IP address and some indication of the length of the mask. Length means the number of leftmost contiguous mask bits that are set to one. So network 172.16.0.0 255.255.0.0 can be represented as 172.16.0.0/16. CIDR also depicts a more hierarchical Internet architecture, where each domain takes its IP addresses from a higher level. This allows for the summarization of the domains to be done at the higher level. For example, if an ISP owns network 172.16.0.0/16, then the ISP can offer 172.16.1.0/24, 172.16.2.0/24, and so on to customers. Yet, when advertising to other providers, the ISP only needs to advertise 172.16.0.0/16.

Hybrid Fiber Coaxial Introduction

Hybrid fibre-coaxial (HFC) is a telecommunications industry term for a broadband network which combines optical fiber and coaxial cable. It has been commonly employed globally by cable TV operators since the early 1990s. See diagram below for a typical architecture for an HFC Network. The fiber optic network extends from the cable operators' master headend, sometimes to regional headends, and out to a neighbourhood's hubsite, and finally to a fiber optic node which serves anywhere from 25 to 2000 homes. A master headend will usually have satellite dishes for reception of distant video signals as well as IP aggregation routers. Some master headends also house telephony equipment for providing telecommunications services to the community. A regional or area headend/hub will receive the video signal from the master headend and add to it the Public, Educational and/or Governmental (PEG) channels as required by local franchising authorities or insert targeted advertising that would appeal to a local area. The various services are encoded, modulated and upconverted onto RF carriers, combined onto a single electrical signal and inserted into a broadband optical transmitter. This optical transmitter converts the electrical signal to a downstream optically modulated signal that is sent to the nodes. Fiber optic cables connect the headend or hub to optical nodes in a point-to-point or star topology, or in some cases, in a protected ring topology. The optical portion of the network provides a large amount of flexibility. If there are not many fiber optic cables to the node, wavelength division multiplexing can be utilised to combine multiple optical signals onto the same fiber. Optical filters are used to combine and split optical wavelengths onto the single fiber. For example, the downstream signal could be on a wavelength at 1310nm and the return signal could be on a wavelength at 1550nm. There are also techniques to put multiple downstream and upstream signals on a single fiber by putting them at different wavelengths. The coaxial portion of the network connects 25 to 2000 homes (500 is typical) in a tree-and-branch configuration off of the node. Radio frequency amplifiers are used at intervals to overcome cable attenuation and passive losses of the electrical signals caused by splitting or "tapping" the coaxial cable. Trunk coaxial cables are connected to the optical node and form a coaxial backbone to which smaller distribution cables connect. Trunk cables also carry AC power which is added to the cable line at usually either 60V or 90V by a power supply and a power inserter. The power is added to the cable line so that trunk and distribution amplifiers do not need an individual, external power source. From the trunk cables, smaller distribution cables are connected to a port of the trunk amplifier to carry the RF signal and the AC power down individual streets. If needed, line extenders, which are smaller distribution amplifiers, boost the signals to keep the power of the television signal at a level that the TV can accept. The distribution line is then "tapped" into and used to connect the individual drops to customer homes. These taps pass the RF signal and block the AC power unless there are telephony devices that need the back-up power reliability provided by the coax power system. The tap terminates into a small coaxial drop using a standard screw type connector known as an F connector. The drop is then connected to the house where a ground block protects the system from stray voltages. Depending on the design of the network, the signal can then be passed through a splitter to multiple TVs. If too many TVs are connected, then the picture quality of all the TVs in the house will go down requiring the use of a "drop" or "house" amplifier.

Transport over HFC network

By using frequency division multiplexing, an HFC network may carry a variety of services, including analogue TV, digital TV (SDTV or HDTV), Video on demand, telephony, and high-speed data. Services on these systems are carried on Radio Frequency (RF) signals in the 5 MHz to 1000 MHz frequency band. The HFC network can be operated bi-directionally, meaning that signals are carried in both directions on the same network from the headend/hub office to the home, and from the home to the headend/hub office. The forward-path or downstream signals carry information from the headend/hub office to the home, such as video content, voice and internet data. The return-path or upstream signals carry information from the home to the headend/hub office, such as control signals to order a movie or internet data to send an email. The forward-path and the return-path are actually carried over the same coaxial cable in both directions between the optical node and the home. In order to prevent interference of signals, the frequency band is divided into two sections. In countries that have traditionally used NTSC System M, the sections are 52 MHz to 1000 MHz for forward-path signals, and 5 MHz to 42 MHz for return-path signals. Other countries use different band sizes, but are similar in that there is much more bandwidth for downstream communication instead of upstream communication. As detailed above, much more of the frequency band is dedicated to the forwardpath than the return-path. Traditionally much more information is sent in the forwardpath due to video content only needing to be sent to the home, so the HFC network is structured to be non-symmetrical, meaning that one direction has much more datacarrying capacity than the other direction. Years ago, the return-path was only used for some control signals to order movies, etc., which required very little bandwidth. As additional services have been added to the HFC network, such as internet data and telephony, the return-path is being utilised more.

Architecture

Softswitch
A softswitch is a central device in a telecommunications network which connects telephone calls from one phone line to another, entirely by means of software running on a computer system. This work was formerly carried out by hardware, with physical switchboards to route the calls. A softswitch is typically used to control connections at the junction point between circuit and packet networks. A single device containing both the switching logic and the switching fabric can be used for this purpose; however, modern technology has led to a preference for decomposing this device into a Call Agent and a Media Gateway. Softswitch is the concept of separating the network hardware from network software. In traditional circuit switched networks, hardware and software is not independent. Circuit switched networks rely on dedicated facilities for inter-connection and are designed primarily for voice communications. The more efficient packet based networks use the Internet Protocol (IP) to efficiently route voice and data over diverse routes and shared facilities. he Call Agent takes care of functions including billing, call routing, signalling, call services and so on and is the 'brains' of the outfit. A Call Agent may control several different Media Gateways in geographically dispersed areas over a TCP/IP link. The Media Gateway connects different types of digital media stream together to create an end-to-end path for the media (voice and data) in the call. It may have interfaces to connect to traditional PSTN networks like DS1 or DS3 ports (E1 or STM1 in the case of non-US networks), it may have interfaces to connect to ATM and IP networks and in the modern system will have Ethernet interfaces to connect VoIP calls. The call agent will

instruct the media gateway to connect media streams between these interfaces to connect the call - all transparently to the end-users. The softswitch generally resides in a building owned by the telephone company called a central office. The central office will have telephone trunks to carry calls to other offices owned by the telecommunication company and to other telecommunication companies (aka the Public Switched Telephone Network or PSTN). Looking towards the end users from the switch, the Media Gateway may be connected to several access devices. These access devices can range from small Analog Telephone Adaptors (ATA) which provide just one RJ11 telephone jack to an Integrated Access Device (IAD) or PBX which may provide several hundred telephone connections.

Feature server as a part of softswitch

The feature server, often built into a call agent/softswitch, is the functional component that provides call-related features. Capabilities such as call forwarding, call waiting, and last call return, if implemented in the network, are implemented in the feature server. The feature server works closely with the call agent, and may call upon the media server to provide these services. These features do not require the subscriber to explicitly request them but tend to be triggered within the call handling logic.

Next Generation Network

A Next Generation Network (NGN) is a packet based network able to provide Telecommunication services in which service related functions are independent from underlying transport technologies. Next generation networking (NGN) is a broad term to describe key architectural evolutions in telecommunication core and access networks that will be deployed over the next 510 years. The general idea behind NGN is that one network transports all information and services (voice, data, and all sorts of media such as video) by encapsulating these into packets, like it is on the Internet. NGNs are commonly built around the Internet Protocol, and therefore the term "all-IP" is also sometimes used to describe the transformation toward NGN.

NGN is a brand new network integrating voice, data, fax, and video services. It is an open and integrated network architecture which includes interfaces to support management functions such as service provisioning, billing, fault removal etc. NGN is commonly associated with voice (a vision for the future of packet-based voice networks), as part of the evolution from TDM circuit switched voice of today. The telephone network (both fixed + mobile) and the Internet are likely to converge into what some people refer to as Next Generation Networks or NGN. To support NGN, There is ongoing standardization to provide integration and interoperability of IP-based and PSTN network services and applications. From a practical perspective, NGN involves three main architectural changes that need to be looked at separately:

In the core network, NGN implies a consolidation of several (dedicated or overlay) transport networks each historically built for a different service into one core transport network (often based on IP and Ethernet). It implies amongst others the migration of voice from a circuit-switched architecture (PSTN) to VoIP, and also migration of legacy services such as X.25, Frame Relay (either commercial migration of the customer to a new service like IP VPN, or technical emigration by emulation of the "legacy service" on the NGN). In the wired access network, NGN implies the migration from the dual system of legacy voice next to xDSL setup in local exchanges to a converged setup in which the DSLAMs integrate voice ports or VoIP, making it possible to remove the voice switching infrastructure from the exchange In cable access network, NGN convergence implies migration of constant bit rate voice to CableLabs PacketCable standards that provide VoIP and SIP services. Both services ride over DOCSIS as the cable data layer standard.

In an NGN, there is a more defined separation between the transport (connectivity) portion of the network and the services that run on top of that transport. This means that whenever a provider wants to enable a new service, they can do so by defining it directly at the service layer without considering the transport layer - i.e. services are independent of transport details. Increasingly applications, including voice, tend to be independent of the access network (de-layering of network and applications) and will reside more on end-user devices (phone, PC, set-top box). Next Generation Networks are based on Internet technologies including Internet Protocol (IP) and Multiprotocol Label Switching (MPLS). At the application level, Session Initiation Protocol (SIP) seems to be taking over from ITU-T H.323. Initially H.323 was the most popular protocol, though its popularity decreased in the "local loop" due to its original poor traversal of Network address translation (NAT) and firewalls. For this reason as domestic VoIP services have been developed, SIP has been more widely adopted. However in voice networks where everything is under the control of the network operator or telco, many of the largest carriers use H.323 as the protocol of choice in their core backbones. So really SIP is a useful tool for the "local loop" and H.323 is like the "fiber backbone". With the most recent changes introduced for H.323, it is now possible for H.323 devices to easily and consistently traverse NAT and firewall devices, opening up the possibility that H.323 may again be looked upon more favorably in cases where such devices encumbered its use previously. Nonetheless, most of the telcos are extensively

researching and supporting IP Multimedia Subsystem (IMS), which gives SIP a major chance of being the most widely adopted protocol. For voice applications one of the most important devices in NGN is a Softswitch a programmable device that controls Voice over IP (VoIP) calls. It enables correct integration of different protocols within NGN. The most important function of the Softswitch is creating the interface to the existing telephone network, PSTN, through Signalling Gateways and Media Gateways. However, the Softswitch as a term may be defined differently by the different equipment manufacturers and have somewhat different functions. One may quite often find the term Gatekeeper in NGN literature. This was originally a VoIP device, which converted (using gateways) voice and data from their analog or digital switched-circuit form (PSTN, SS7) to the packet-based one (IP). It controlled one or more gateways. As soon as this kind of device started using the Media Gateway Control Protocol, the name was changed to Media Gateway Controller (MGC). A Call Agent is a general name for devices/systems controlling calls. The IP Multimedia Subsystem (IMS) is a standardized NGN architecture for an Internet media-services capability defined by the European Telecommunications Standards Institute (ETSI) and the 3rd Generation Partnership Project (3GPP).

Benefits of Migration onto NGN

One Integrated network for voice and data. Horizontally integrated layer on common transport layer based on packet technology which can be shared by different services. Control layer is separated from transport layer to provide all types of emerging multimedia services and to keep at the same time existing services provided by ATM, IP/LPMS bases services. Operators are shifting onto NGN taking into account the following 1. Data Services Transported through a multi service Packet

Networks: To become capable of integrating a variety of data services which

today run on IP, ATM, FR. 2. Consolidation of Different Overlay Networks: To utilize the existing Circuit switch networks instead of dismantling them.

3. Consolidated Control and Reduced Operating Cost:

NGN is a centrally managed network, which in turn reduces the operating cost drastically, Moreover all new services can be managed centrally as the same are provision from one platform using less resources, thus reducing operating cost.

NGN Characteristics
Packet-based transfer

Broadband capabilities with end-to-end QoS and transparency Interworking with legacy networks via open interfaces generalized mobility Unrestricted access by users to different service providers

Architecture

NGN Network Architecture

Operational Elements Business continuity

required to maintain ongoing dominant services and customers that require carrier-grade service Flexibility to incorporate existing new services and react quickly to the ones that appear on real time (main advantage of IP mode) Profitability to allow feasible return on investments and in the best practices market values Survivability to allow service assurance in case of failures and external unexpected events Quality of Service to guarantee the Service Level Agreements for different traffic mixes, conditions and overload. Interoperabilty across networks to allow to carry end to end services for flows in different network domains

Network Elements Packet based networks

Trend is to use IP based networks over various transport possibilities (ATM, SDH, WDM) IP based networks must offer guarantees of Quality of Service (QoS) regarding the real time characteristics of voice, video and multimedia

Access Gateways
Allows the connection of subscriber lines to the packet network Converts the traffic flows of analogue access (Pots) or 2 Mb/s access devices into packets Provides subscriber access to NGN network and services

Trunking Gateways
Allows interworking between classical TDM telephony network and Packet-based NGN networks, Converts TDM circuits/ trunks (64kbps) flows into data packets, and vice versa

Softswitch/MGC
Referred to as the Call Agent or Media Gateway Controller (MGC). Provides the service delivery control within the network in charge of Call Control and handling of Media Gateways control (Access And/or Trunking) via H.248 protocol Performs signalling gateway functionality or uses a signalling gateway for Interworking with PSTN N7 signalling network Provides connection to Intelligent Network /applications servers to offer the same services as those available to TDM subscribers

Application Server (AS):

A unit that supports service execution, e.g. to control Call Servers and NGN special resources (e.g. media server, message server).

H.248 Protocol
Known also as MEGACO: standard protocol, defined by ITU-T, for signalling and session management needed during a communication between a media gateway, and the media gateway controller managing it H.248/MEGACO allows to set up, keep, and terminate calls between multiple endpoints as between telephone subscribers using the TDM

SIP
Session Initiation Protocol in order to handle call establishment, maintenance and termination from packet mode terminals.

Signalling Gateway (SG):

A unit that provides signalling conversion between the NGN and the other networks (e.g. STP in SS7).

ENUM
Electronic NUMbering: Protocol that allows to establish a correspondance between the traditional telephone numbering (E.164 ) and the network addresses related to the packet mode networks ( RFC 2916 "E.164 number and DNS" IETF).

MPLS
Multiprotocol Label Switch or protocol that assigns labels to information packets in order to allow the node routers to treat and route flows in the network paths according to established priority for each category.

CAC
Call Acceptance Control function in order to accept/reject traffic in the network that allows guarantee of QoS for services with a Service Level Agreement

BGP
Border Gateway Protocol to negotiate flow routing procedures and capacities across different NGN network domains

CAPACITY OF NGN MEDIA GATEWAYS MEDIA GATEWAY

UNIVERSAL MEDIA GATEWAY ACCESS GATEWAY

MAXIMUM CAPACITY
256 E1s 5000 Subscribers

NGN Development

Why Selecting NGN Networks

There are many factors involved in the selection of NGN Networks which are as follows

Factors such as:

Lower port cost Open platform Reduced space and power requirements New revenue streams Improved competitive position Lower operating costs