Network fundamental

IP,IPv6 Routing Protocols, Internet protocols version six, IPv6

noreply@blogger.com (Utsav Basu) — Mon, 10 Aug 2009 05:41:34 PDT

Source :

Deploying IPv6 Networks
By Ciprian Popoviciu, Eric Levy-Abegnoli, Patrick Grossetete
...............................................
Publisher: Cisco Press
Pub Date: February 10, 2006

IPv6 Packet Format
IPv6 Routing Protocols

Numerous IPv4 routing protocols (RPs) are available for finding routes between networks, and almost every one of them has an IPv6 correspondent or extension: Routing Information Protocol next-generation (RIPng), Open Shortest Path First version 3 (OSPFv3), Intermediate System-to-Intermediate System (IS-IS), Enhanced Interior Gateway Routing Protocol (EIGRP), and Border Gateway Protocol (BGP). So far, IPv6 has brought few innovations to the IP routing paradigm. There are still interior gateway protocols (IGPs) and exterior gateway protocols (EGPs), distance vectorbased and link-state-based routing protocol algorithms, and so on.

The concept of the autonomous system, defined as a set of networks controlled by a common administrative entity, remains unchanged with the introduction of IPv6 RPs. The same autonomous system (and autonomous system number [ASN]) will route both IPv4 and IPv6. IPv6 IGPs, used to exchange routes within the autonomous system, are namely RIPng, OSPFv3, IS-IS for IPv6, and EIGRP for IPv6. Only BGP4 is available to exchange IPv6 routes between autonomous systems. Multiprotocol extensions provide support in BGP4 for IPv6 routing.

The requirements for IGPs and EGPs are quite different, in terms of routing table size, number of supported routers, convergence time, security, routing policy, and so forth. For that reason, they use different algorithms and mechanisms, which also affect the type of information they exchange and store. IGPs use distance vector and link state, whereas BGP uses the path vector RP algorithm. The following table represents RP taxonomy, and highlights their IPv6 correspondent. For more details on how to choose the RP, refer to Top-Down Network Design, Second Edition, by Priscilla Oppenheimer.

Table 4-1. Taxonomy of Routing Protocols
Deployment Domain	Algorithm	RP	Scalability	Convergence Time	Metric	IPv6 Version
Interior gateway protocol (IGP)	Distance vector	RIP	15 hops	Slow	Hop count	RIPng
		EIGRP	1000s routers	Quick (via DUAL algorithm)	Bandwidth, delay, reliability, load	EIGRP for IPv6
	Link state	OSPF	1000s routers (100s/area)	Quick (via LSAs and HELLO)	Cost (function of bandwidth on Cisco routers)	OSPFv3
		IS-IS	1000s routers (100s/area)	Quick (via LSPs)	Configured host, delay, expense	IS-IS for IPv6
Exterior gateway protocol (EGP)	Distance vector	EGP			Integer <=255
	Path vector	BGP AF=IPv4	1000s routers	Slow (via UPDATE)	Function of path attributes and other configurable factors	BGP AF=IPv6

The rest of this section briefly reviews existing unicast RP technologies. The next section reviews each of the available IPv6 RPs and provides configuration examples. Then the last two sections cover the topic of multihoming and deployment aspects, respectively.

IP Mobility, IP, IPv6, IP version 6, Internet protocol

noreply@blogger.com (Utsav Basu) — Mon, 10 Aug 2009 05:42:02 PDT

Source :

Deploying IPv6 Networks
By Ciprian Popoviciu, Eric Levy-Abegnoli, Patrick Grossetete
...............................................
Publisher: Cisco Press
Pub Date: February 10, 2006

IP Mobility

The Internet has become so pervasive that no matter where you are, you can plug your computer into a wall, or attach to a wireless LAN, and, after a while, you will be able to communicate. Is not this mobility? Well, not quite.

That type of "mobility" is achieved by getting a new IP address within the network of attachment and losing all sessions bound to the previous IP address. This might be acceptable for corporate users moving from work to home, but can be much more cumbersome for road warriors, and it can be a showstopper for IP telephony.

Mobile IP provides a network layer for hosts that enables them to maintain the same IP address no matter where they are in the Internet, and keep receiving traffic as they move.

"Advanced ServicesIPv6 Mobility," MIPv6 is compared to MIPv4. Even though MIPv4 is a mature and deployable technology, it faces limitations because of the nature of IPv4. At the same time, IPv6 mobility is considered as one potential enabler for IPv6. The number of IP-enabled devices and the need for any-to-any communications among them is driving requirements that IPv4 cannot easily satisfy, and it is opening opportunities for IPv6. By integrating functionalities designed for Mobile IPv4 into standard IPv6 protocols, and by leveraging existing IPv6 capabilities, MIPv6 has built up a MIP model that is much more compelling than its IPv4 counterpart.

It must be noted that enhancements to mobility are largely taking place in IPv6 related working groups, even though a fraction gets retrofitted into the IPv4 standards. Although MIPv6 has benefited greatly from its MIPv4 parent, it is now the driver of the evolution of IP mobility, and it is widely expected to be a foremost steering force for IPv6 deployments.

In terms of deployment, it must be considered that IP mobility enables new flows, which impact the wireless infrastructure: Telephony over IP demands a higher level of coverage, latency, and QoS enforcement, whereas peer to peer imposes always-on reachability and multimedia capabilities.

The application of the MIP and NEtwork MObility (NEMO) standards is not limited to hosts and routers that actually roam around the Internet as a usual behavior. Sales of consumer routers are plummeting. At the moment, they are related to IPv4 NAT operations. With IPv6, it can be expected that people will deploy unmanaged yet globally addressable networks at home. NEMO support by the home gateways would enable a service provider to deploy preprovisioned devices, and could save hundreds of thousands of network-renumbering operations per year as customers move from one home to the next.

At the core, MIP builds dynamic tunnels, and NEMO exchanges routes over those tunnels. In a way, this is a revamping of the traditional model of the core where BGP routers exchange the bulk of the Internet routes over peering tunnels. But whereas the model of the Internet is designed for fixed, aggregated routes that are locally injected and slowly distributed throughout its fabric, MIP and NEMO techniques enable a new model where routes are projected where and when they are needed, on-demand; this opens to a new level of hierarchy for the fine-grained mobile routes, and a new order of scalability for the Internet.

But the Internet of today is not fully ready for IP mobility. Even if IPv6 can exist over an IPv4 fabric as a transitional method, a significant number of improvements must be made to cope with the latency of the protocol and enable multimedia interactive applications such as voice calls and video.

Data Cabling Rules,Reliable Cabling, Poor Cabling Cost

noreply@blogger.com (Utsav Basu) — Sat, 01 Aug 2009 21:43:15 PDT

The Golden Rules of Data Cabling,The Importance of Reliable Cabling,The Legacy of Proprietary Cabling Systems, Cabling and the Need for Speed, Cable Design , Data Communications 101, Speed Bumps: What Slows Down Your Data, The Future of Cabling Performance. Learn how to Cabling, Know that better than the best.

Source: Cabling:The Complete Guide to Network Wiring (Third Edition)
Author : David Barnett, David Groth, Jim McBee.

“Data cabling! It’s just wire. What is there to plan?” the newly promoted programmer turned MIS-director commented to Jim. The MIS director had been contracted to help the company move its 750-node network to a new location. During the initial conversation, the director had a couple of other “insights”:
●He said that the walls were not even up in the new location, so it was too early to be talking about data cabling.
●To save money, he wanted to pull the old Category 3 cabling and move it to the new location. (“We can run 100Base-TX on the old cable.”)
●He said not to worry about the voice cabling and the cabling for the photocopier tracking system; someone else would coordinate that. Jim shouldn’t have been too surprised by the ridiculous nature of these comments. Too few people understand the importance of a reliable, standards-based, flexible cabling system. Fewer still understand the challenges of building a high-speed network. Some of the technical problems associated with building a cabling system to support a high-speed network are comprehended only by electrical engineers. And many believe that a separate type of cable should be in the wall for each application (PCs, printers, terminals, copiers, etc.). Data cabling has come a long way in the past 20 years. This chapter discusses some of the basics of data cabling, including topics such as:

●The golden rules of data cabling
●The importance of reliable cabling
●The legacy of proprietary cabling systems
●The increasing demands on data cabling to support higher speeds
●Cable design and materials used to make cables
●Types of communications media
●Limitations that cabling imposes on higher-speed communications
●The future of cabling performance

You are probably thinking right now that all you really want to know is how to install cable to support a few 10Base-T workstations. Words and phrases such as attenuation,crosstalk,twisted pair,modular connectors, and multi-mode optical-fiber cable may be completely foreign to you. Just as the world of PC LAN's and WANs has its own industry buzzwords, so does the cabling business. In fact, you may hear such an endless stream of buzzwords and foreign terminology that you’ll wish you had majored in electrical engineering in college. But it’s not really that mysterious and, armed with the background and information we’ll provide, you’ll soon be using cable speak like a cabling professional.

The Golden Rules of Data Cabling

Listing our own golden rules of data cabling is a great way to start this chapter and the book. If your cabling is not designed and installed properly, you will have problems that you can’t even imagine. From our experience, we’ve become cabling evangelists, spreading the good news of proper cabling. What follows is our list of rules to consider when planning structuredcabling systems:

●Networks never get smaller or less complicated.
●Build one cabling system that will accommodate voice and data.
●Always install more cabling than you currently require. Those extra outlets will come in
handy someday.
●Use structured-cabling standards when building a new cabling system. Avoid anything
proprietary!
●Quality counts! Use high-quality cabling and cabling components. Cabling is the foundation of your network; if the cabling fails, nothing else will matter. For a given grade or category of cabling, you’ll see a range of pricing, but the highest prices don’t necessarily mean the highest quality. Buy based on the manufacturer’s reputation and proven performance, not the price.
●Don’t scrimp on installation costs. Even quality components and cable must be installed correctly; poor workmanship has trashed more than one cabling installation.
●Plan for higher speed technologies than are commonly available today. Just because 1000Base-T Ethernet seems unnecessary today does not mean it won’t be a requirement in five years.
●Documentation, although dull, is a necessary evil that should be taken care of while you’re setting up the cabling system. If you wait, more pressing concerns may cause you to ignore it.

The Importance of Reliable Cabling

We cannot stress enough the importance of reliable cabling. Two recent studies vindicated our evangelical approach to data cabling. The studies showed:

●Data cabling typically accounts for less than 10 percent of the total cost of the network infrastructure.
●The life span of the typical cabling system is upwards of 16 years. Cabling is likely the second most long-lived asset you have (the first being the shell of the building).
●Nearly 70 percent of all network-related problems are due to poor cabling techniques and cable-component problems.

Of course, these were facts that we already knew from our own experiences. We have spent countless hours troubleshooting cabling systems that were nonstandard, badly designed, poorly documented, and shoddily installed. We have seen many dollars wasted on the installation of additional cabling and cabling infrastructure support that should have been part of the original installation. Regardless of how you look at it, cabling is the foundation of your network. It must be reliable!

The Cost of Poor Cabling

The costs that result from poorly planned and poorly implemented cabling systems can be staggering. One company that had recently moved into a new office space used the existing cabling, which was supposed to be Category 5 cable. Almost immediately, 100Mbps Ethernet network users reported intermittent problems.

These problems included exceptionally slow access times when reading e–mail, saving documents, and using the sales database. Other users reported that applications running under Windows 98 and Windows NT were locking up, which often caused them to have to reboot their PC. After many months of network annoyances, the company finally had the cable runs tested. Many cables did not even meet the minimum requirements of a Category 5 installation, and other cabling runs were installed and terminated poorly.

Network Fundamental and its Components

noreply@blogger.com (Utsav Basu) — Mon, 27 Jul 2009 20:52:34 PDT

Source : Microsoft Encyclopedia of Networking Second edition

What Is Networking?

In the simplest sense, networking means connecting computers so that they can share files, printers, applications, and other computer-related resources. The advantages of networking computers are fairly obvious:
● Users can save their important files and documents on a file server. This is more secure than storing them on workstations because a file server can be backed up in a single operation.
● Users can share a network printer, which costs much less than having a locally attached printer for each user’s computer.
● Users can share groupware applications running on application servers, which enables users to share documents, send messages, and collaborate directly.
● The job of administering and securing a company’s computer resources is simplified since they are concentrated on a few centralized servers. The above definition of networking focuses on the basic goals of networking computers together: increased manageability, security, cost-effectiveness, and efficiency over non-networked systems. However, we could also focus our discussion on the different types of networks, including
● Personal area networks (PANs), once the stuff of science fiction but rapidly becoming a reality as the mobile knowledge workers of today carry around an array of cell phones, Personal Digital Assistants (PDAs), pagers, and other small devices
● Local area networks (LANs), which can range from a few desktop workstations in a Small Office/Home Office (SOHO) to several thousand workstations and dozens of servers deployed throughout dozens of buildings on a university campus or in an industrial park
● Metropolitan area networks (MANs), which span an urban area and are generally run by telcos and other service providers to provide companies with high-speed connectivity between branch offices and with the Internet
● Wide area networks (WANs), which might take the form of a company’s head office linked to a few branch offices or an enterprise spanning several continents with hundreds of offices and subsidiaries
● The Internet, the world’s largest network and the “network of networks” On the other hand, we could also focus on the different networking architectures in which these various types of networks can be implemented, including
● Peer-to-peer networking, which might be implemented in a workgroup consisting of computers running Microsoft Windows 98 or Windows 2000 Professional
● Server-based networking, which might be based on the domain model of Windows NT, the domain trees and forests of Active Directory directory service in Windows 2000, or another architecture such as Novell Directory Services (NDS) for Novell NetWare
● Terminal-based networking, which might be the traditional host-based mainframe environment; the UNIX X Windows environment; the terminal services of Windows NT Server 4 Enterprise Edition; Windows 2000 Advanced Server; or Citrix MetaFrame Or we could look at the various networking technologies used to implement these architectures, including
● LAN technologies such as Ethernet, Token Ring, Fiber Distributed Data Interface (FDDI), Fast
Ethernet, Gigabit Ethernet (GbE), and the emerging 10G Ethernet (10GbE)
● WAN technologies such as Integrated Services Digital Network (ISDN), T-carrier leased lines, X.25, frame relay, Asynchronous Transfer Mode (ATM), Synchronous Optical Network (SONET), Digital Subscriber Line (DSL), and metropolitan Ethernet
● Wireless communication technologies such as the wireless LAN (WAN) standards 802.11a and
802.11b, and the consumer wireless technologies HomeRF and Bluetooth
● Cellular communication systems such as Time Division Multiple Access (TDMA), Code Division
Multiple Access (CDMA), Global System for Mobile Communications (GSM), and the emerging
3G cellular communication standards In addition, we could consider the hardware used to
implement these different networking technologies, including

● LAN devices such as repeaters, concentrators, bridges, hubs, Ethernet switches, and routers
● WAN devices such as modems, ISDN terminal adapters, Channel Service Units (CSUs), Data Service Units (DSUs), packet assembler/disassemblers (PADs), frame relay access devices (FRADs), multiplexers (MUXes), and inverse multiplexers (IMUXes)
● Equipment for organizing, protecting, and troubleshooting LAN and WAN hardware, such as racks, cabinets, surge protectors, line conditioners, uninterruptible power supplies (UPSs), KVM switches, and cable testers
● Cabling technologies such as coaxial cabling, twinax cabling, twisted-pair cabling, fiber-optic cabling, and associated equipment such as connectors, patch panels, wall plates, and splitters
● Unguided media technologies such as infrared communication, wireless cellular networking, and satellite networking, along with their associated hardware
● Data storage technologies such as redundant array of independent disks (RAID), network-attached storage (NAS), and storage area networks (SANs) along with their associated hardware, plus various enabling technologies, including Small Computer System Interface (SCSI) and Fibre Channel Or we could talk about various technologies that enhance the reliability, scalability, security, and manageability of computer networks, including
● Technologies for implementing network security, including firewalls, proxy servers, and virtual private networking (VPN), and such devices as smart cards and firewall appliances
● Technologies for increasing availability and reliability of access to network resources, such as clustering, caching, load balancing, Layer 7 switching, and terminal services
● Network management technologies such as Simple Network Management Protocol (SNMP), Remote Network Monitoring (RMON), Web-Based Enterprise Management (WBEM), Common Information Model (CIM), and Windows Management Instrumentation (WMI) Returning to a more general level, networking can also be thought of as the various standards that underlie the
different networking technologies and hardware mentioned above, including
● The Open Systems Interconnection (OSI) networking model from the International Organization for Standardization (ISO)
● The G-series, H-series, I-series, T-series, V-series, and X-series standards from the International Telecommunication Union (ITU)
● Project 802 of the Institute of Electrical and Electronics Engineers (IEEE)
● The Requests for Comment (RFC) series from the Internet Engineering Task Force (IETF)
● Various standards developed by the World Wide Web Consortium (W3C), the Frame Relay Forum, the ATM Forum, the Gigabit Ethernet Alliance, and other standards organizations Networking protocols deserve special attention in any definition of the word networking. These protocols include:
● LAN protocols such as NetBEUI, Internetwork Packet Exchange/Sequenced Packet Exchange
(IPX/SPX), Transmission Control Protocol/Internet Protocol (TCP/IP), and AppleTalk
● WAN protocols such as Serial Line Internet Protocol (SLIP), Point-to-Point Protocol (PPP), Point-to- Point Tunneling Protocol (PPTP), and Layer 2 Tunneling
Protocol (L2TP)
● Protocols developed within mainframe computing environments, such as Systems Network Architecture (SNA), Advanced Program-to-Program Communications (APPC), Synchronous Data Link Control (SDLC), and High-level Data Link Control (HDLC)
● Routing protocols such as the Routing Information Protocol (RIP), Interior Gateway Routing Protocol (IGRP), Open Shortest Path First (OSPF) Protocol, and Border Gateway Protocol (BGP)
● Internet protocols such as the Hypertext Transfer Protocol (HTTP), File Transfer Protocol (FTP), Network News Transfer Protocol (NNTP), and the Domain Name System (DNS)
● Electronic messaging protocols such as X.400, Simple Mail Transfer Protocol (SMTP), Post Office Protocol version 3 (POP3), and Internet Mail Access Protocol version 4 (IMAPv4)
● Directory protocols such as X.500’s Directory Access Protocol (DAP) and the Lightweight Directory Access Protocol (LDAP)
● Security protocols such as Password Authentication Protocol (PAP), Challenge Handshake Authentication Protocol (CHAP), Windows NT LAN Manager (NTLM) Authentication, Kerberos, IP Security Protocol (IPsec), Secure Sockets Layer (SSL), and public key cryptography standards and protocols
● Serial interface standards such as RS-232, RS-422/ 423, RS-485, V.35, and X.21
We could dig still deeper and discuss the fundamental engineering concepts that underlie the various networking technologies and services previously discussed, including
● Impedance, attenuation, shielding, near-end crosstalk (NEXT), and other characteristics of cabling and other transmission systems
● Signals and how they can be multiplexed using time-division, frequency-division, statistical, and other multiplexing techniques
● Transmission parameters including bandwidth, throughput, latency, jabber, jitter, backbone, handshaking, hop, dead spots, dark fiber, and late collisions
● Balanced vs. unbalanced signals, baseband vs. broadband transmission, data communications equipment (DCE) vs. data terminal equipment (DTE), circuit switching vs. packet switching, connection-oriented vs. connectionless communication, unicast vs. multicast and broadcast, pointto- point vs. multipoint links, direct sequencing vs. frequency hopping methods, and switched virtual circuit (SVC) vs. permanent virtual circuit (PVC) We could also talk about the different types of providers of networking services, including
● Internet service providers (ISPs), application service providers (ASPs), and integrated communications providers (ICPs)
● Telcos or local exchange carriers (LECs), including both Regional Bell Operating Companies (RBOCs) and competitive local exchange carriers (CLECs), that offer such popular broadband services as Asymmetric Digital Subscriber Line (ADSL) and High-bit-level Digital Subscriber Line (HDSL) through their central office (CO) and local loop connection
● Inter-exchange carriers (IXCs) that provide popular WAN services such as dedicated leased lines and frame relay for the enterprise (large companies)
● Local loop alternatives including cable modems, fixed wireless, and satellite networking companies We could also list the various software technologies vendors have developed that make computer networking both useful and possible, including
● Network operating systems such as Windows, Novell NetWare, UNIX, and Linux
● Specialized operating systems such as Cisco Systems’ Internetwork Operating System (IOS), which runs on Cisco routers, and the variant of IOS used on Cisco’s Catalyst line of Ethernet switches
● Directory systems such as Microsoft Corporation’s domain-based Active Directory, Novell Directory Services (NDS), and various implementations of X.500 and LDAP directory systems
● File systems such as NTFS file system (NTFS) on Windows platforms and distributed file systems such as the Network File System (NFS) developed by Sun Microsystems for the UNIX platform
● Programming languages and architectures for developing distributed computing applications, such as the C/C++ and Java languages, Microsoft’s ActiveX and Sun’s Jini technologies, component technologies such as Distributed Component Object Model (DCOM) and COM+, inter process communication (IPC) technologies such as Remote Procedure Calls (RPCs) and named pipes, and Internet standards such as the popular Hypertext Markup Language (HTML) and the Extensible Markup Language (XML) family of standards
● Tools for integrating networking technologies in heterogeneous environments, such as Gateway Services for NetWare (GSNW), Services for Macintosh, Services for UNIX on the Windows 2000 platforms, and Microsoft Host Integration Server, all of which provide connectivity with mainframe systems On an even deeper level, we could focus on the various administration tools for managing networking hardware, platforms, services and protocols, including
● The Microsoft Management Console (MMC) and its various snap-ins in the Windows 2000 and
Windows .NET Server platforms
● The various ways routers and network appliances can be administered using Telnet, terminal programs, and the universal Web browser interface
● Popular TCP/IP command-line utilities such as arp, ping, ipconfig, traceroute, netstat, nbtstat, finger, and nslookup
● Platform-specific command-line utilities such as various Windows commands used for automating common administration tasks
● Cross-platform scripting languages that can be used for system and network administration, including JavaScript, VBScript, and Perl We could also look at various enterprise applications widely used in networked environments, including
● Enterprise Resource Planning (ERP) and Customer Relationship Management (CRM) platforms
● Enterprise Information Portal (EIP) and Enterprise Knowledge Portal (EKP) platforms
● The Microsoft .NET Enterprise Server family of applications that includes Microsoft Application Center Server, BizTalk Server, Commerce Server,Exchange Server, Host Integration Server, Internet Security and Acceleration Server, Mobile Information Server, and SQL Server I think that you can see by now that we could go on and on, slowly unpeeling our answer to the question “What is networking?” like the many layers of an onion. And it is pretty obvious by now that there is more to networking than just hubs and cables! In fact, the field of computer networking today is almost overwhelming in its breadth and complexity, and one could spend a lifetime studying only one small aspect of the subject. This has not always been the case. Let’s take a look now at how the field of computer networking has reached the amazing point where it is today.

How to Setup a LAN

noreply@blogger.com (Utsav Basu) — Thu, 16 Jul 2009 20:12:21 PDT

Check how to set a lan between two or more computers, full tricks to lan set up information, Configuration the LAN was not so much easy before that so keep continue reading to set a lan

SETTING UP YOUR OWN LAN

LAN Overview

This chapter shows how to set up a local network in your home or office (if you already have a functioning network, feel free to skip to the next chapter). In the next few pages, we’ll take a look at:

· The basics of network hardware

· Basic hardware requirements for your local network

· Installing the hardware and setting up a local peer-to-peer network

· Inspecting and changing network settings

A. NETWORK HARDWARE BASICS

Cables. We recommend Category 5 cables for new users. Officially called Ethernet 10/100BaseT,
they’re the most common type of network cable and provide a good upgrade path should you need it. Cat 5 allows either 10- or 100-megabyte communication. These terms have simple meanings, so don’t let them put you off:

· The “10” or “100” in 10/100BaseT refers to network connection speed—i.e., 10 Megabits or 100 Megabits per second. Most networks actually top out at less, though most users would never know.

· The “T” in BaseT refers to the wire type, twisted-pair, which consists of pairs of thin wires twisted around each other. It also refers to the connector, commonly called an RJ-45, which resembles a bigger and wider telephone connector.

· “Base” means that the cable is used for baseband (i.e., simple, single frequency) rather than broadband (multiplex or analog) networks. Cables can be purchased in different lengths and often different colors. They come with a male RJ-45 plug at each end. Cards and hubs have female RJ-45 jacks. Network Cards. A wide variety of network cards—officially called Network Interface Cards and nicknamed NICs—is available. Most do at least an adequate job. If you’re a novice networker, the primary things to look for are:

· Connection Jack. Be sure the NIC’s jack matches the type of cable you’re using. If you’re using 10BaseT cable, for instance, the NIC you buy should have an RJ-45 compatible connector.

· Plug and Play compatibility. This feature allows Windows 95/98 to automatically configure the card, saving you a lot of time in the process.

· Interrupt Addresses. Interrupts on any machine are at a premium, so you’ll want to determine
which ones the NIC has available. Generally, the more you pay, the more latitude you’ll have. ISA-bus cards are usually fast enough for a 10BaseT network; if you’re running 100BaseT you’ll
probably want to go with PCI-bus card for speed. If you’ve only got one interrupt left and must add two cards, use two PCI-bus network cards; part of the PCI spec is that cards can share Interrupts.

Hubs.

Ethernet is a standardized way of connecting computers together to create a network. A hub is an ethernet device used in conjunction with 10BaseT and 100BaseT cables. The cables run from the network’s computers to ports on the hub. Using a hub makes it easier to move or add computers, find and fix cable problems, and remove computers temporarily from the network (if they need to be upgraded, for instance). Hubs are available in most computer stores. It’s probably a good idea to buy one with more ports than you need, just in case your network expands. Look for:

· A connection jack compatible with your cabling.

· A cascading jack which allows you to add an additional hub later, if necessary, without replacing
the entire unit.

· Lights on the front. These can be useful when you’re trying to diagnose network connection problems.

B. HARDWARE FOR YOUR SPECIFIC LAN REQUIREMENTS

The kind of hardware you use depends on the kind of access and/or modem you’re using. If you’re using dial-up access you’ll need:

· One network card for each computer.

· One hub.

· A cable for each connection to the hub.

If you’re using cable modem, DSL modem or direct access you’ll need:

· One network card for each computer.

· One additional network card to connect to the modem (your WinProxy machine receives two cards, one for the modem and one for the local network).

· One hub.

· A cable for each hub connection.

· An additional cable for the connection from the computer to the modem. If the modem is the type that connects directly to the hub, make this last cable a cross-over cable instead and you’ll still be able to connect directly to the network card as shown. Before you rush out and buy a ransom’s worth of network hardware, however, take a few moments to draw a topography—a diagram which shows the relation between the network’s various components. Doing so lessens the chance that you’ll buy unnecessary cables or forget to buy a hub. Let’s look at a very simple topography. Assuming that you already have Internet access through an ISP, you’re probably connected to the Internet in this manner:

Now let’s look at the topography for a simple LAN. The network shown here—the number of client machines can be far greater, of course—is the standard configuration for most setups, including dialup access and cable-modem access:

As you can see, only one computer—the WinProxy computer—has a modem. The other computers are connected to each other and to the WinProxy computer by a device called a hub (more on this later). The computer using the modem and receiving the WinProxy installation must be a Windows95/98 or Windows NT machine. Other computers on the local network can be any kind—including Macs, Unix boxes, and WfWG3.11—as long as they’re capable of “speaking” TCP/IP. Once you’ve drawn your network topography, including all components, make a list of everything you need.

Notes

1. Many cable modem providers insist on installing the cable modem card

themselves, and may insist upon using their own card. Before purchasing your

own cables and cards, check to see what the provider’s policy is.

2. If you have only two computers, it’s possible to save the expense of a

hub by connecting them back-to-back. To do so, run a cross-over cable directly

from one network card to the network card on the other machine. IP addressing

will still be done as described here

C. HARDWARE INSTALLATION/SETTING UP THE PEER-TOPEER

NETWORK

The best way to install an NIC is to simply follow the manufacturer’s directions. Win95/98 usually finds a new card when it starts up and then configures it for you. If it doesn’t, consult the directions that came with the card. Run a cable between each card and the hub (except for the external network card if you have a cable modem setup). Although you can probably get away with plugging/unplugging a cable from a card while the computer is running, it’s safer to do it when the computer is turned off. You can usually plug or unplug from the hub at any time.

You’ll need at least one protocol assigned to each card once it’s installed. Choose NetBEUI (NetBios Extended User Interface) at a minimum; you can have others as well. There isn’t any problem with having multiple protocols on your local network. You’ll need the TCP/IP protocol later in order to run WinProxy, but it’s not needed now when setting up a basic peer-to-peer network. Set up your basic network first, get it working, and we’ll add TCP/IP later on. During the card setup, you’ll be prompted for certain settings. If not already installed, be sure to add for each machine:

· Client for Microsoft Networks

· File and Printer Sharing

You can make changes to your settings at any time in the future. You must reboot the computer
for the changes to take effect.

D. INSPECTING AND CHANGING NETWORK SETTINGS

At this point, let’s double-check the computer network setup at Control Panel/Networks. In the window under the Configuration Tab, you’ll see a list of adapters and protocols. A typical setup is represented by a couple of small computer-shaped icons, one captioned Client for Microsoft Networks, and the other File and Printer Sharing. You’ll also see small green icons, similar in shape to a network card—one for each network card, and one for the Dial-Up Adapter (the Dial-Up Adapter counts as a network connection, with its own set of addresses and protocols). Finally, you’ll see a series of wire-and-node icons, each listing a different protocol-and-adapter combination, written in a form something like NetBEUI Æ NE2000 Compatible Card.

If you haven’t already added Client for Microsoft Networks, do so now:

· Highlight an adapter.

· Click through the path Add/Client/Microsoft/Client for Microsoft Networks.

To add a protocol capability to a network card:

· Highlight the network card.

· Click through the path Add/Protocol/Microsoft/Your Protocol. Click on the Identification Tab, where you’ll see three entry boxes titled:

Computer name: A name assigned by you to a computer (each computer on the network should have a unique name). Avoid punctuation marks. These names are frequently used in network configurations, and you’ll save confusion later by assigning distinctive names now. Old486 is a good name if you only have one 486 computer, but if you have several, assign them names like PapaBear, MamaBear, etc. NetBEUI uses this name to find things so it can perform its networking magic. You’ll sometimes see this computer name referred to as “the NetBios name.”

Workgroup Name: A group name you can assign to all the computers on your network (or you can use the default).

Computer Description: A caption that gives users on your local network information about an individual computer. An example: Maria’s Computer

Security Alert The designated protocol will usually be assigned exactly as you’ve requested. (In Windows 95 and 98, however, Microsoft assigns the NetBEUI protocol to all network adapters when you assign it to any single network adapter). If you don’t want that protocol in the other locations, highlight each one you don’t want and click Remove.

A Final Word on Your LAN

Congratulations! You now possess a working local network. You can see the other computers, move files between them, and print documents. To prepare for WinProxy and the Internet, you’ll need to add the TCP/IP protocol to each of the computers on your local network. You’ll learn how to do so in the next chapter. Once that’s done, it’s on to WinProxy!

ADDING TCP/IP TO YOUR NETWORK

TCP/IP Overview

The easiest way to install and configure WinProxy is to first add TCP/IP—the language spoken by WinProxy and the Internet—to your local network. This chapter covers the following topics:

1. Protocols and Addressing

2. Double-Checking Your Installed Network

3. Installing TCP protocol on your computers

4. Assigning IP addresses

5. Testing TCP/IP connectivity

A. FIRST THINGS FIRST: PROTOCOLS AND ADDRESSING Protocols. In networking terms the word “protocol” refers to the accepted standards or rules for the way data is transferred between computers and over the Internet. When everybody uses the same rules, it all works. There are many protocols in use. The three commonly used by local networks are NetBEUI, IPX/SPX, and TCP/IP.

NetBEUI is an acronym which stands for NetBios Extended User Interface. NetBEUI is a networking standard well suited for small networks and is easy to set up. It is also non-routable; since it uses computer names to find its way around, it can’t find distant computers.

IPX/SPX is Novell network’s version of IP addressing, used on Novell NetWare networks for both small and large systems. It works on Novell networks, but not between different types of networks (as TCP/IP will). TCP/IP, the language of the Internet, can be used on any size network. Data is sent over the network in chunks called packets. TCP (Transmission Control Protocol) is the protocol for packets of data sent over the wires. IP (Internet Protocol) is the addressing method used to get these packets to and from the right place. It is a routable protocol, designed to find distant computers. Some carefully-defined address groups are designated as intentionally non-routable; we’ll be using one of these to set up TCP/IP on your local network in the next chapter.

Network Addresses. These addresses may be assigned manually by the user, or automatically by another computer. They’re called static (i.e., fixed) assignments when assigned by the user, because they stay the same over time. When assigned automatically by computer, they’re known as dynamic (i.e., changing) assignments. If you connect via a dial-up connection, you’ll probably have a dynamic IP assignment to your Dial-Up Adapter. Your ISP assigns a different IP address to your Dial-Up Adapter each time you connect. If connecting with a cable modem, you’ll most likely have to make a static IP assignment for your Internet connection. Once this assignment is made, the IP address will not change. In addition, you’ll also have the choice of static or dynamic addresses on most of your networked computers. You can either set static IP addressing information yourself or have WinProxy make dynamic IP assignments for you. Addresses are not assigned to the computer itself, though people often speak that way as a convenient shorthand. The addresses are actually assigned to each network connection. The computer on which WinProxy will be installed, for instance, will have two network connections: an internal connection to the rest of your computers, and an external connection to your Internet Service Provider, or ISP.

In “Internet speak,” any machine with a network address is called a Host. For most simple TCP/IP systems, each host is a computer, and each computer is a host. The IP address is a 32-bit address, subdivided into four fields. Although it’s a binary number, it’s usually written in decimal form—222.5.83.47, for example. Each field can have a value from 0 through 255. However, since the end values are used for special purposes, the actual range available is from 1 to 254. What this boils down to for you, the user, is this: when entering an IP address, use only numbers between 1 and 254 in that last field.

Associated with the IP address is the subnet mask. This mask tells the computer which part of the address is unique to that machine, and which part is the general network address. Subnet masks allow you to accomplish many esoteric addressing capabilities; however, for most simple networks the subnet mask of 255.255.255.0 is the best and easiest choice. When you use this mask, the numbers in the final field of the IP address are unique to each computer, and the preceding three fields define the network address. To learn more about the intricacies of subnet masks. Some specific IP address ranges are reserved for special uses. We’ll discuss these later when setting up IP addressing on your local network. Network addresses reserved for testing or for local networks are 10.x.x.x, 90.x.x.x, 172.16-31.x.x and 192.168.x.x. These addresses all share a crucial distinction: routing computers on the Internet will not route these numbers. Since they are perfectly good numbers on a local network, but cannot be routed across the Internet, using them adds security to your local network.

Parts of a TCP packet are fields that specify the source and destination ports. These are 16-bit fields, and can thus specify more than 65 thousand ports. You’ll see many references to ports when interfacing your local net to the Internet. Ports 1 through 1024 are set aside for specific uses. Each Internet protocol has a standard port assigned to its use (e.g., Port 25 to send mail, Port 119 for news groups). In many cases, things can be easier to follow if you consider a port designation to be part of the address; some software even allows you to specify an IP address and port combination in the same statement.

B. DOUBLE-CHECKING YOUR INSTALLED NETWORK

Before going further, let’s double-check to be sure you have a basic network installed. At this point, your network should look like this:

· Your computers are connected via a working Ethernet network.

· One of the computers has an Internet connection, and is using Windows 95/98 or NT. That computer gets the WinProxy installation and will be known as the WinProxy computer.

· You already have some network protocols installed, including NetBEUI, and your computers already have NetBios names. The NetBios name of each computer can be found at Control Panel/Network/Identification/Computer Name. If your network doesn’t match these specifications, please bring it into line, using Chapter 3 to guide you, before attempting to install TCP Protocol and WinProxy. On the other hand, if you do have a basic network, read on!

C. INSTALLING TCP/IP PROTOCOL ON YOUR COMPUTERS

All communication between the client applications and WinProxy, and between WinProxy and the Internet, use TCP/IP protocols. Thus, the first thing you must do is add the TCP protocol and IP addresses to the network’s computers. As you proceed, pay attention to the dictates of the following three connection types:

1. The external WinProxy connection to the Internet. The type of IP address used—dynamic (commonly used for standard modems) or static (commonly used for cable modems)—is dictated by the ISP to which you connect and the type of service it provides.

2. The internal WinProxy connection. This connection must be a static IP assignment, and it must be assigned by you. Two reasons exist for a static assignment. First, some client applications must have a single, known address for the proxy server; second, the static assignment is used by WinProxy as a starting place for its DHCP assignments when providing tcp/ip assignments to your other computers.

3. The client computer network connections. These connections can be either dynamic or static. If they’re dynamic, WinProxy automatically makes all IP assignments and settings—the preferred method when using the WinProxy 3.0 Install Wizard. If they’re static, you must enter IP settings for each client computer. We recommend dynamic assignments for new users. Several protocols can co-exist on a local network, and you’ll usually need to have more than one. One protocol is sufficient on the connection to the Internet, and for security reasons you should have only TCP/IP. Let’s proceed. To install TCP.

1. On the machine receiving the WinProxy installation, click Control Panel/Network/Configuration. You’ll see a list of installed new components, and there should be listings for a Dial-Up adapter and a LAN adapter (exact wording varies). Look under LAN adapter to see if you have TCP installed—if it is, the listing will read something like TCP/IP ® LAN Adapter. Again, the exact wording varies.

2. If TCP/IP isn’t listed, click through this path: LAN Adapter/Add/Protocol/ Microsoft/TCPIP/
OK. That’s it! You’ll be prompted to restart, finishing the installation. Do so if you like, or you can wait until completing the next step before restarting.

3. Return to the initial screen. Look under Dial-Up Adapter to see if you have TCP/IP installed. If not, click through this path: DialUp Adapter/Add/ Protocol/Microsoft/TCP-IP/OK. When prompted to restart the computer, do so.

4. Add the TCP/IP protocol to each client machine (unless it’s already installed). The process is the same: in Control Panel/Networks look for a TCP/IP ® LAN Adapter line, adding the TCP/IP
protocol to the LAN adapter if it isn’t already installed.

For Client Machines Only After completing Step 4, take a quick look at any dial-up adapters. If any are installed and have the TCP protocol assigned, look under Properties to ensure that the dial-up adapter does not have the option Assign a specific IP address selected. It should be set to Obtain an IP address automatically. This will save you trouble down the road.

D. ASSIGNING IP ADDRESSES TO ALL NETWORK COMPUTERS

Each computer must be assigned a unique IP address. Strictly speaking, an IP address is assigned to each network connection, but it’s convenient to speak of a “machine address.” If you set your client computers to Obtain an IP address automatically (see the boxed note immediately above), WinProxy takes care of all of these settings for you. We recommend using the 90.0.0.x series of addresses on your local network. You’ll reap three major benefits by doing so:

· Your setup will match the numbers used for diagrams and instructions on the WinProxy website.

· You’ll find it much easier to follow explanations and trouble-shoot your network problems should the need arise.

· You’ll add to the security of your local computers by using this non-routing series on your local
network.

Now let’s proceed to assigning IP addresses.

1. First, let’s assign an IP address to the WinProxy machine. To do so, follow this path:

Control Panel/Network Configuration/TCP/IP/LANAdapter/Properties. Bring the IP Address Tab to the front. Click Specify an IP Address and enter an IP address and subnet mask. We recommend 90.0.0.1 and 255.255.255.0, as shown in the screen. You shouldn’t need to make any changes on other tabs for this basic installation.

2. Use the method shown above to install an IP address on each client machine. It’s easiest to use a sequential series such as 90.0.0.2, 90.0.0.3 and so on. Each computer gets a subnet mask of 255.255.255.0. Each IP address on your local network must be unique, and you can only vary the number in the final group—in other words, don’t change the 90.0.0 portion of the address.

3. If you’ll be using a dial-up connection to an Internet provider, the dial-up adapter does not get a specific IP assignment. Set it to Obtain an IP Address Automatically. The IP address for this connection will be dynamically assigned by the ISP each time you connect. These addresses come from a pool, and will probably (but not necessarily) be different each time you connect.

4. If, instead, you’ll be using a direct connection to your Internet provider (as many cable modems do), the network card connected to the modem should be assigned the IP address and subnet mask specified by your ISP for your individual Internet connection. Remember: you must have two network cards on this machine—one for the direct external connection to your provider and one for the internal connection to the rest of the computers on your local network.
The network card connecting to the rest of your local network retains the IP assignment it received in Step 1, above. At the conclusion of your installation, click through to WinProxy/Advanced Properties/General/ Multiple IP. While there, check to see that the IP number assigned to your Internet connection is defined as an external connection, and the IP number assigned to your local network is defined as an internal connection.

E. TESTING TCP/IP CONNECTIVITY

Now that you’ve added TCP/IP to all your computers, let’s run a test to determine if Network Neighborhood is up and running properly. If it is, you’ll know that the hub and cables are working correctly. We’ll use Ping for our test. It’s a simple tool included in Windows 95/98 and NT that allows easy checking of TCP/IP connectivity. First, open a DOS box (Start/Program/MS-DOS Prompt in Windows 95/98, and Start/Program/Command Prompt in Windows NT) and type the word ping. You’ll see a list of. commands and command syntax. If you’re on, say, client machine 90.0.0.2, you can check your connectivity to the WinProxy machine by typing in its IP address (90.0.0.1) after you type the word ping. If TCP/IP is properly set up on both machines you’ll get several lines that say Reply from 90.0.0.1…, as shown in the screen below. If you get no reply, something is wrong with the protocol installation of the IP address on one (or both) machines.

This series of three tests, run on each machine with a communications problem, will probably help isolate the problem:

1. Ping 127.0.0.1 to ensure that your tcp/ip software is working.

2. Ping yourself to ensure that the card is working.

3. Test to see that you can communicate with another machine.

· To run the first test (pinging the loopback address), type ping 127.0.0.1 at the DOS prompt. This verifies that the software TCP/IP stack on that machine is working and that the TCP protocol has been assigned (bound) to the card. The loopback address is specifically designated for such tests and doesn’t generate any actual network traffic. A failure at this point would implicate the software. If that’s the case, consider re-installing Winsock from your Windows CD-ROM, or download and install the latest Winsock from Microsoft.

· Now ping the IP address of the WinProxy computer, verifying that the card is working and IP addressing is correctly configured on that machine. If you discover a problem at this point, check to see that your network card is working properly. In Windows 95/98, go to Control Panel\System\Device Manager to see if there is a yellow exclamation point or question mark on your network card. If there is, click Drivers, and then choose View Resources to determine if Windows reports a conflict—e.g., an interrupt conflict. If so, you may be able to resolve the conflict by assigning an unused interrupt. If not, try reinstalling the card.

· Ping the IP address of another machine on your network. To work properly, the configuration must be correct on both machines. A problem at this stage usually indicates an IP addressing error. You’ve probably violated one of the basic IP rules, perhaps assigning the same number to two different machines, assigning a number outside the allowed range, or simply mis-typing an address. Check and double-check the assigned addresses. If you get a response such as request timed out, it means that ping did not reach (or return from) the other machine. Look for misconfigured IP addresses or unplugged hubs. If your response is something like destination unreachable, then ping didn’t know how to follow through on your request. You might get this response if, for example, you pinged an address with a different set of network fields. Look for misnamed nets or misconfigured subnet masks.

USER’S CHECKPOINT: If everything works except the last test (pinging another computer) an old proxy installation may be interfering. Proxy software that requires installation of software components on client machines as well as on the proxy server can cause tcp/ip communication problems. This software must be removed from each machine for proper tcp/ip communication.

If there seems to be a problem with a network card, go to Control Panel/ System/Device Manager/View Devices by Type. Look under Network Adapters. If you see a yellow exclamation point or question mark over the adapter, the system is having a problem with that adapter. Use the Win95/98 wizards to help track down problems. If you upgraded from Windows 95 to Windows 98, your network card drivers are probably out of date. Download new drivers made specifically for Windows 98 from the manufacturer’s web site.

Dial UP Technology

noreply@blogger.com (Utsav Basu) — Thu, 11 Jun 2009 00:41:07 PDT

Dial-up Technology

Dialup is simply the application of the Public Switched Telephone Network (PSTN) to carry data on behalf of the end user. It involves a customer premises equipment (CPE) device sending the telephone switch a phone number to direct a connection to. The AS3600, AS5200, AS5300, and AS5800 are all examples of routers that have the capability to run a PRI along with banks of digital modems. The AS2511, on the other hand, is an example of a router that communicates with external modems. Since the time of Internetworking Technologies Handbook, 2nd edition, the carrier market has continued to grow, and there have been demands for higher modem densities. The answer to this need was a higher degree of interoperation with the telco equipment and the refinement of the digital modem: a modem capable of direct digital access to the PSTN. This has allowed the development of faster CPE modems that take advantage of the clarity of signal that the digital modems enjoy. The fact that the digital modems connecting into the PSTN through a PRI or a BRI can transmit data at more than 53 K using the V.90 communication standard attests to the success of the idea.

A Short Dialup Technology Background

Dialup technology traces its origins back to the days of the telegraph. Simple signals being sent across an extended circuit were created manually by tapping contacts together to turn the circuit either on or off. In an effort to improve the service, Alexander Graham Bell invented the telephone in 1875 and changed communication forever. Having the capability to send a voice across the line made the technology more accessible and attractive to consumers. By 1915, the Bell system stretched from New York to San Francisco. Demand for the service drove technological innovations, which led to the first transatlantic phone service in 1927 via radio signal. Other innovations along the way included. microwave stations that started connecting American cities in 1948, integrated digital networks to improve the quality of service, and communication satellites, which went into service in 1962 with the launch of Telstar 1. By 1970, more than 90 percent of American homes had telephone service. In 1979 the modulator-demodulator (modem) was introduced, and dialup networking was born. The early modems were slower and subject to proprietary communication schemes. Early uses of modems were for intermittent point-to-point WAN connections. Often, the call would come into a regular phone at a data center. An operator would hear modem tones and place the handset onto a special cradle that was the modem. In the late 1980s, the ITU-T began setting up V-series recommendations to standardize communications between both data communications equipment (DCE) and data terminal equipment (DTE). Early standards included these:

• V.8—Standardized the method that modems use to initially determine the V-series modulation at which they will communicate. Note that this standard applies only to the communication session between the two DCE devices. This was later updated with V.8bis, which also specified some of the communication standards between the DTE devices going over the DCE’s connection.
• V.21, V.23, V.27ter, V.29—Defined 300, 600/1200, 2400/4800, and 9600 baud communications, respectively.
• V.25, V.25bis, V.25ter—Served as a series of standards for automated dialing, answering, and control. Modems increased greatly in sophistication in the late 1980s. This was due in part to the breakup of the Bell system in 1984. With the client premises equipment in the hands of free enterprise, competition spurred on the development of speedier connections. More recent standards include these:
• V.32bis, V.34, V.90—Standardized 14400, 33600, and up to 56000 baud communication speeds.
• V.110—Allowed an asynchronous DTE device to use an ISDN DCE (terminal adapter). The first access servers were the AS2509 and the AS2511. The AS2509 could support 8 incoming connections using external modems, while the AS2511 could support 16. The AS5200 was introduced with 2 PRIs and could support 48 users using digital modems—this represented a major leap forward in technology. Modem densities have increased steadily, with the AS5300 supporting four and then eight PRIs. The AS5800 was later introduced to fill the needs of carrier class installations needing to handle dozens of incoming T1s and hundreds of user connections A couple of outdated technologies bear mentioning in a historical discussion of dialer technology. 56
Kflex is an older (pre-V.90) 56 K modem standard that was proposed by Rockwell. Cisco supports version 1.1 of the 56 Kflex standard on its internal modems, but it recommends migrating the CPE modems to V.90 as soon as possible. Another outdated technology is the AS5100. The AS5100 was a joint venture between Cisco and a modem manufacturer. The AS5100 was created as a way to increase modem density through the use of quad modem cards. It involved a group of AS2511s built as cards that were inserted into a backplane shared by quad modem cards, and a dual T1 card. Today dialup is still used as an economical alternative (depending on the connection requirements) to dedicated connectivity. It has important uses as backup connectivity, in case the primary lines go down. Dialup also offers the flexibility to create dynamic connections as needed.

Dialup Connectivity Technology

This section provides information from various dialup options. Also included are advanced options for dialup connectivity and various dialup methods.

Plain Old Telephone Service

The regular phone lines used in voice calls are referred to as Plain old telephone service (POTS). They are ubiquitous, familiar, and easy to obtain; local calls are normally free of charge. This is the kind of service that the phone network was built on. Sounds carried over this service are sampled at a rate of 8000 times per second (using 8 bits per sample) in their conversion to digital signals so that sound can be carried on a 64 kbps channel at acceptable levels.The encoding and decoding of voice is done by a piece of telco gear called a CODEC. The CODEC was needed to allow backward-compatibility with the old analog phones that were already in widespread use when the digital network was introduced. Thus, most phones found in the home are simple analog devices. Dialup connectivity across POTS lines has historically been limited to about 33,600 bps via modem—often referred to as V.34 speeds. Recent improvements have increased the speed at which data can be sent from a digital source to a modem on a POTS line, but using POTS lines on both ends of the connection still results in V.34 connectivity in both directions.

Basic Rate Interface

Intended for home use, this application of ISDN uses the same copper as a POTS line, but it offers direct digital connectivity to the telephone network. A special piece of equipment known as a terminal adapter is required (although, depending on the country, it may be integrated into the router or DCE device). Always make sure to check—the plug used to connect to the wall socket looks the same whether it’s the S/T or U demarcation point. Normally, a Basic rate interface (BRI) interface has two B (bearer) channels to carry data, and one D (delta) channel to carry control and signaling information. Local telephone carriers may have different plans to suit local needs. Each B channel is a 64 K line. The individual 64 K channels of the telephone network are commonly referred to as digital service 0 (DS0). This is a common denominator regardless of the types of services offered, as will be shown later in this chapter. The BRI interface is a dedicated connection to the switch and will remain up even if no calls are placed. The T1/E1 line is designed for use in businesses. T1 boasts 24 TDM channels run across a cable with 2 copper pairs. E1 offers 32 channels, although 1 is dedicated to frame synchronization. As is the case with the BRI, the T1/E1 connection goes directly into the telco switch. The connection is dedicated, so like a BRI, the T1/E1 remains connected and communicating to the switch all the time—even if there are no active calls. Each of the channels in the T1/E1 is just a B channel, which is to say that it’s a 64-K DS0. The T1/E1 is also referred to as digital service 1 (DS1). The North American T1 uses frames to define the timing between individual channels. For T1s, each frame has 24 9-bit channels (8 bits of data, 1 bit for framing). That adds up to 193 bits per frame. So, at
8000 of those per second, the T1 is carrying 1.544 Mbps between the switch and the customer premises equipment (CPE). The E1 similarly uses frames for timing, but the E1 uses 32 8-bit channels for a 256-bit frame. Again at the 8000 Hz rate, the channel yields 2.048 Mbps of traffic between the switch and the CPE. Most of the world uses the E1. Depending on the region, various line code and framing schemes will have to be used for the CPE and the switch to understand each other. For example, in North America, the encoding scheme most often seen is called binary 8 zero substitution (B8ZS), and the most common framing done is extended super frame (ESF). The telco through which the T1/E1 service is purchased must indicate which line code and framing should be used. For dialup purposes, there are two types of T1/E1: Primary Rate Interface (PRI) and channel associated signaling (CAS). PRI and CAS T1/E1s are normally seen in central locations that receive calls from remote sites or customers.

Primary Rate Interface

T1 Primary rate interface (PRI) service offers 23 B channels at 64 kbps at the cost of one D-channel (the 24th channel) for call signaling. Using NFAS to allow multiple PRIs to use a single D channel can minimize this disadvantage. E1 PRI service allows 30 channels, but it uses the 16th channel for ISDN signaling. The PRI service is an ISDN connection. It allows either voice-grade (modem) or true ISDN calls to be made and received through the T1/E1. This is the type of service most often seen in access servers because it fosters higher connection speeds.

Channel Associated Signaling

T1 Channel associated signaling (CAS) lines have 24 56K channels—part of each channel is borrowed for call signaling. This type of service is also called robbed-bit signaling. The E1 CAS still uses only the 16th channel for call signaling, but it uses the R2 international standard for analog call signals. CAS is not an ISDN interface; it allows only analog calls to come into the access server. This is often done to allow an access server to work with a channel bank, and this scenario is seen more commonly in South America, Europe, and Asia,sends a call into a channel that isn’t expecting it, the switch will get back a message indicating that the channel isn’t available. An access server must maintain state information on its lines and be prepared to coordinate inward and outward calls with the switch.

Modems

From a terminology standpoint, a modem is considered data communication equipment (DCE), and the device using the modem is called data terminal equipment (DTE). As indicated earlier, modems must adhere to a number of communication standards to work with other modems: Bell103, Bell212A, V.21, V.22, V.22bis, V.23, V.32, V.32bis, V.FC, and V.34, to name a few. These standards reflect a dual analog conversion model,Notice that the signal goes through only one analog conversion. Because the conversion is done on the client’s side, traffic generated by the client modem is limited to V.34 speeds. The traffic coming from the access server is not subject to the noise problems that an analog conversion would introduce, so it can be sent at much higher speeds. Thus, the client can receive data at v.90 speeds but can send data at only V.34 speeds.

PPP -

PPP bears mentioning because it is so vital to the operation of dialup technologies. Until PPP came along in 1989 (RFC 1134—currently up to RFC 1661), dialup protocols were specific to the protocol being used. To use multiple protocols, it was necessary to encapsulate any other protocols within packets of whatever protocol the dialup link was running. Many of the proprietary link methods (such as SLIP) didn’t even have the capability to negotiate addressing. Fortunately, PPP does this and many more things with flexibility and extensibility. PPP connection establishment happens in three phases: Link Control Protocol (LCP), authentication, and Network Control Protocol (NCP).

LCP-

LCP is the lowest layer of PPP. Because PPP does not follow a client/server model, both ends of the point-to-point connection must agree on the negotiated protocols. When negotiation begins, each of the peers wanting to establish a PPP connection must send a configure request (CONFREQ). Included in the CONFREQ are any options that are not the link default. These often include maximum receive unit, async control character map, authentication protocol, and the magic number. At this stage, the peers negotiate their authentication method and indicate whether they will support PPP multilink. In the general flow of LCP negotiations, there are three possible responses to any CONFREQ:

1. A configure-acknowledge (CONFACK) must be issued if the peer recognizes the options and agrees to the values seen in the CONFREQ.
2. A configure-reject (CONFREJ) must be sent if any of the options in the CONFREQ are not recognized (such as some vendor-specific options) or if the values for any of the options have been explicitly disallowed in the configuration of the peer.
3. A configure-negative-acknowledge (CONFNAK) must be sent if all the options in the CONFREQ are recognized, but the values are not acceptable to the peer. The two peers continue to exchange CONFREQs, CONFREJs, and CONFNAKs until each sends a CONFACK, until the dial connection is broken, or until one or both of the peers indicates that the negotiation cannot be completed.

Authentication

Authentication is an optional phase, but it is highly recommended on all dial connections. In some
instances, it is a requirement for proper operation—dialer profiles, being a case in point. The two principal types of authentication in PPP are the Password Authentication Protocol (PAP) and the Challenge Handshake Authentication Protocol (CHAP), defined by RFC 1334 and updated by RFC 1994. When discussing authentication, it is helpful to use the terms requester and authenticator to distinguish the roles played by the devices at either end of the connection, although either peer can act in either role. Requester describes the device that requests network access and supplies authentication information; the authenticator verifies the validity of the authentication information and either allows or disallows the connection. It is common for both peers to act in both roles when a DDR connection is being made between routers. PAP is fairly simple. After successful completion of the LCP negotiation, the requester repeatedly sends its username/password combination across the link until the authenticator responds with an acknowledgment or until the link is broken. The authenticator may disconnect the link if it determines that the username/password combination is not valid. CHAP is somewhat more complicated. The authenticator sends a challenge to the requester, which then responds with a value. This value is calculated by using a “one-way hash” function to hash the challenge and the CHAP password together. The resulting value is sent to the authenticator along with the requester’s CHAP host name (which may be different from its actual host name) in a response message. The authenticator reads the host name in the response message, looks up the expected password for that host name, and then calculates the value that it expects the requester to send in its response by performing the same hash function the requester performed. If the resulting values match, the authentication is successful. Failure should lead to a disconnection. By RFC standards, the authenticator can request another authentication at any time during the connection.

NCP-

NCP negotiation is conducted in much the same manner as LCP negotiation with CONFREQs, CONFREJs, CONFNAKs, and CONFACKs. However, in this phase of negotiation, the elements being negotiated have to do with higher-layer protocols—IP, IPX, bridging, CDP, and so on. One or more of these protocols may be negotiated. Refer to the following RFCs for more detail on their associated protocols:

• RFC 1332 “IP Control Protocol”
• RFC 1552 “IPX Control Protocol”
• RFC 1378 “AppleTalk Control Protocol”
• RFC 1638 “Bridging Control Protocol”
• RFC 1762 “DECnet Control Protocol”
• RFC 1763 “VINES Control Protocol”

A Couple of Advanced Considerations

The Multilink Point-to-Point Protocol (MLP, RFC 1990) feature provides a load-balanced method for splitting and recombining packets to a single end system across a logical pipe (also called a bundle) formed by multiple links. Multilink PPP provides bandwidth on demand and reduces transmission latency across WAN connections. At the same time, it provides multivendor interoperability, packet fragmentation with proper sequencing, and load calculation on both inbound and outbound traffic. The Cisco implementation of multilink PPP supports the fragmentation and packet sequencing specifications in RFC1717. Multilink PPP works over the following interface types (single or multiple):

• Asynchronous serial interfaces
• BRIs
• PRIs

Multichassis multilink PPP (MMP), on the other hand, provides the additional capability for links to terminate at multiple routers with different remote addresses. MMP can also handle both analog and digital traffic. This functionality is intended for situations in which there is a large pool of dial-in users, and a single access server cannot provide enough dial-in ports. MMP allows companies to provide a single dialup number to their users and to apply the same solution to analog and digital calls. This feature allows Internet service providers, for example, to allocate a single ISDN rotary number to several ISDN PRIs and not have to worry about whether a user’s second link is on the same router. MMP does not require reconfiguration of telephone company switches.

AAA

Another technology that should be mentioned because of its importance is Authentication, Authorization, and Accounting (AAA). The protocols used in AAA can be either TACACS or RADIUS. These two protocols were developed in support of a centralized method to keep track of users and accesses made on a network. AAA is employed by setting up a server (or group of servers) to centrally administer the user database. Information such as the user’s password, what address should be assigned to the user, and what protocols the user is allowed to run can be controlled and monitored from a single. workstation. AAA also has powerful auditing capabilities that can be used to follow administratively important trends such as connection speeds and disconnect reasons. Any medium or large dialup installation should be using AAA, and it’s not a bad idea for small shops, either.

Dialup Methods

Most routers support automated methods for dynamic links to be connected when traffic that needs to get to the other end arrives. Cisco’s implementation is called dial-on-demand routing (DDR). It provides WAN connectivity on an economical, as-needed basis, either as a primary link or as backup for a nondial serial link. At its heart, DDR is just an extension of routing. Interesting packets are routed to a dialer interface that triggers a dial attempt. Each of the concept’s dialer interface and interesting traffic bear explanation.

What’s a Dialer?

The term dialer has a few meanings, depending on the specifics of the configuration, but in general, it refers to the interface where the routing is actually happening. This is the interface that knows the address and phone number where the traffic is supposed to go. When looking at the routing table, the dialer interface should be the interface referenced for the next hop to reach the network on the other side. The dialer interface does not have to be the physical interface that is doing the dialing, but it can be made so by placing the configuration command dialer in-band in a physical interface. Thereafter, the interface becomes a dialer. For example, an async interface is not a dialer by default, but placing the configuration command dialer in-band in the async interface causes dialer behavior on that interface. For example, calls received by that async interface after applying the command will have an idle timeout applied to the connection from then on. An example of a physical interface that is also a dialer by default would be the BRI interface. Beyond making physical interfaces into dialers, there are interfaces called dialer interfaces. These are logical interfaces that call upon real interfaces to place calls. The advantage of using a dialer interface is flexibility. A group of potential DDR links can share a handful of BRI interfaces. Dialer interface configuration comes in two flavors: dialer map-based (sometimes referred to as legacy DDR) and dialer profiles. Which method you use depends on the circumstances under which you need dial connectivity. Dialer map-based DDR was first introduced in IOS Version 9.0; dialer profiles were introduced in IOS Version 11.2.

Interesting Traffic

The term interesting is used to describe packets or traffic that will either trigger a dial attempt or, if a dial link is already active, reset the idle timer on the dialer interface. For a packet to be considered interesting, it must have these characteristics:

• The packet must meet the “permit” criteria defined by an access list.
• The access list must be referenced by the dialer–list, or the packet must be of a protocol that is
universally permitted by the dialer–list.
• The dialer-list must be associated with a dialer interface by use of a dialer group.

Packets are never automatically considered to be interesting (by default). Interesting packet definitions must be explicitly declared in a router or access server configuration.

Benefits and Drawbacks

The benefits of dialup are flexibility and cost savings. First, let’s look at why flexibility is important. Intermittent connectivity is most often needed in mobile situations. A mobile workforce needs to be capable of connecting from wherever they are. Phone lines are normally available from wherever business is transacted, so a modem connection is the only reasonable choice for mobile users. In long-distance situations, a user often dials into a local ISP and uses an IPSec-encrypted tunnel going back to a home gateway system that allows access to the rest of the corporate network. In this example, the phone call itself costs nothing, and an account with the local ISP could be significantly less expensive than the long-distance charges that would otherwise be incurred. As another example, a BRI attached at a central office located in an area that offers inexpensive rates on ISDN could have database servers configured to call out to other sites and exchange data periodically. Each site needs only one BRI line, which is significantly less expensive than dedicated links to each of the remote locations. Finally, in the case of a backup link, the savings are seen when the primary link goes down but business continues, albeit slower than normal. Cost savings is a two-edged sword where dialup is concerned, however. The downside of a dialup line is that connection costs for a heavily used line are higher than the price of dedicated connectivity. Going over long distance raises the price even higher. There’s also speed to consider. Dialup connectivity has a strong high-end bandwidth, particularly with the capability to tie channels together using PPP multilink, but dedicated connectivity through a serial port can outperform dialup connections. Another consideration is security. Certainly, any PPP connection should be authenticated, but this presents anyone with the dialup number an opportunity to break into the system. A significant part of any dialup system’s configuration concerns the capability to keep out unwanted guests. The good news is that it can be done, and AAA goes a long way toward dealing with this problem. However, it is a disadvantage to have potential intruders coming in through dialup lines.

Telecommunications Systems

noreply@blogger.com (Utsav Basu) — Sun, 17 May 2009 10:06:12 PDT

Telecommunications Systems

So as we proceed through this material, try not to get frustrated with the constant mix of services, technological discussions, and costing issues. From time to time, we may also introduce some extra technical notes that are for the more technically astute but can be ignored by the novice trying to progress through the industry. As you read about a topic, do so with a focus on systems, rather than individual technologies. We have tried to make these somewhat stand-alone chapters, yet we have also tied them together in bundles of three or four chapters to formulate a final telecommunications system. Do what you must to understand the information, but do not force it as you read. The pieces will all come together throughout the groupings of topics.

What Constitutes a Telecommunications System

A network is a series of interconnections that form a cohesive and ubiquitous connectivity arrangement when all tied together. That sounds rather vague, so let’s look at the components of what constitutes the telecommunications network. The telecommunications network referred to here is the one that was built around voice communications but has been undergoing a metamorphosis for the past two decades. The convergence of voice and data is nothing new; we have been trying to run data over a voice network since the 1970s. However, to run data over the voice network, we had to make the data look like voice. This caused significant problems for the data because the voice network was noisy and error-prone. Reliability was a dream and integrity was unattainable, no matter what the price.

Generally speaking, a network is a series of interconnection points. The telephone companies over the years have been developing the connections throughout the world so that a level of cost-effective services can be achieved and their return on investment (ROI) can be met. As a matter of due course, whenever a customer wants a particular form of service, the traditional carriers offer two answers:

1. It cannot be done technically.
2. The tariff will not allow us to do that!

Regardless what the question happened to be, the telephone carriers were constantly the delay and the limiting factor in meeting the needs and demands for data and voice communications.
In order to facilitate our interconnections, the telephone companies installed wires to the customer’s door. The wiring was selected as the most economical to satisfy the need and the ROI equation. Consequently, the telephone companies installed the least expensive wiring possible. Because they were primarily satisfying the demand for voice communications, they installed a thin wire (26-gauge) to most customers whose locations were within a mile or two from the central office. At the demarcation point, they installed the least expensive termination device (RJ-11), satisfying the standard two-wire unshielded twisted pair communications infrastructure. The position of the demarcation point depended on the legal issues involved. In the early days of the telephone network, the telephone companies owned everything, so they ran the wires to an interface point and then connected their telephone equipment to the wires at the customer’s end. The point here is that the telephone sets were essentially commodity-priced items requiring little special effect or treatment.

When the data communications industry began during the late 1950s, the telephone companies began to charge an inordinate amount of money to accommodate this different service. Functionally, they were in the voice business and not the data business. As a matter of fact, to this day, most telephone companies do not know how to spell the word data ! They profess that they understand this technology, but when faced with tough decisions or generic questions, few of their people can even talk about the services. How sad, they will be left behind if they do not change quickly. New regulations in the U.S., in effect since the divestiture agreement, changed this demarcation point to the entrance of the customer’s building. From there, the customer hooked up whatever equipment was desired. Few people remember that in early 1980, a 2400-bps modem cost $10,000. The items that customers purchase from a myriad of other sources include all the pieces we see during the convergence process. In the rest of the world today, where full divestiture or privatization has not yet taken place, the telephone companies (or PTTs) still own the equipment. Other areas of the world have a hybrid system under which customers might or might not own their equipment. The combinations of this arrangement are almost limitless, depending on the degree of privatization and deregulation. However, the one characteristic that is common in most of the world to date is that the local provider owns the wires from the outside world to the entrance of the customer’s building. This local loop is now under constant attack from the wireless providers offering satellite service, local multipoint distribution services (LMDS), and multi-channel multi-point distribution services (MMDS). Moreover, the CATV companies have installed coaxial cable or fiber, if new wiring has been installed, and they offer the interconnection to business and residential consumers alike. The CLECs have also emerged as formidable foes to the local providers. They are installing fiber to many corporate clients (or buildings) with less expense and long-term write-off issues. The CLECs are literally walking away from the telephone companies and the local loop. Add the xDSL family of products to this equation and the telephone companies are running out of options. The telephone companies today have approximately 15,000 xDSL connections across the U.S., whereas the CLECs have over 200,000. Moreover, the cable TV companies have close to 800,000 cable modems installed for high-speed Internet access. This is where the CATV companies see the convergence taking place. CLECs are also seeing the convergence in the local loop, and with xDSL in their potpourri of offerings, they are actually nudging the local telephone
companies aside.

A Topology of Connections Is Used

In the local loop, the topological layout of the wires has traditionally been a single-wire pair or multiple pairs of wires strung to the customer’s location. Just how many pairs of wires are needed for the connection of a single line set to a telecommunications system and network? The answer (one pair) is obvious. However, other types of services, such as digital circuits and connections, require two pairs. The use of a single or dual pair of wires has been the norm. More recently, the local providers have been installing a fourpair (eight wires) connection to the customer location. The end user is now using separate voice lines, separate fax lines, and separate data communications hookups. Each of these requires a two-wire interface from the LEC. However, if a CATV provider has the technology installed, they can get a single coax to satisfy the voice, fax, data, and highspeed Internet access on a single interface, proving the convergence is rapidly occurring at the local loop. It is far less expensive to install a coax running all services (TV, voice, and data) than multiple pairs of wire, so the topology is a dedicated local connection of one or more pairs from the telephone provider to the customer location or a shared coax from the CATV supplier. This is called a star and/or shared star-bus configuration . The telephone company connection to the customer originates from a centralized point called a central office (CO). The provider at this point might be using a different topology. Either a star configuration to a hierarchy of other locations in the network layout or a ring can be used. The ring is becoming a far more prevalent method of connection for the local Telcos. Although we might also show the ring as a triangle, it is still a functional and logical ring. These star/ring or star/bus combinations constitute the bulk of the networking topologies today. Remember one fundamental fact: the telephone network was designed to carry analog electrical signals across a pair of wires to recreate a voice conversation at both ends. This network has been built to carry voice and does a reasonable job of doing so. Only recently have we been transmitting other forms of communication, such as facsimile, data, and video.

The telephone switch (such as DMS-100 or #5ESS) makes routing decisions based on some parameter, such as the digits dialed by the customer. These decisions are made very quickly and a cross-connection is made in logic. This means that the switch sets up a logical connection to another set of wires. Throughout this network, more or fewer connections are installed, depending on the anticipated calling patterns of the user population. Sometimes there are many connections among many offices. At other times, it can be simple with single connections. The telephone companies have begun to see a shift in their traffic over the past few years. More data traffic is being generated across the networks than ever before. As a matter of fact, 1996 marked the first year that as much data was carried on the network as voice. Since that time, data has continued its escalated growth pattern, whereas voice has been stable.

The Local Loop

Our interface to the telephone company network is the single-line telephone line, which has been installed for decades and is written off after 30 or 40 years. Each subscriber or customer is delivered at least one pair of wires per telephone line. There are exceptions to this rule, such as when the telephone company might have multiple users sharing a single pair of wires. If the number of users demanding telephone service exceeds the number of pairs available, a Telco might offer the service on a party-line or shared set of wires. It is in this outside plant, from the CO to the customer location, that 90 percent of all problems occur. This is not to imply that the Telco is doing a lousy job of delivering service to the customer. In the analog dial-up telephone network, each pair of the local loop is designed to carry a single telephone call to service voice conversations. This is a proven technology that works for the most part and continues to get better as the technologies advance. What has just been described is the connection at the local portion of the network. From there, the local connectivity must be extended out to other locations in and around a metropolitan area or across the country. The connections to other types of offices are then required.

The Telecommunications Network

Prior to 1984, most of the network was owned by AT&T through its local Bell operating telephone companies. A layered hierarchy of office connections was designed around a five-level architecture. Each of these layers was designed around the concept of call completion. The offices were connected together with wires of various types called trunks . These trunks can be twisted pairs of wire, coaxial cables (like the CATV wire), radio (such as microwave), or fiber optics. As the convergence of voice and data networks continues, we see a revisitation to the older technologies as well as the new ones. Fiber is still the preferred medium from a carrier’s perspective. However, microwave radio is making a comeback in our telecommunications systems, linking door-to-door private line services. Carrying voice, data, video, and high-speed Internet access is a natural for a microwave system. Lightbased systems, however, are limited in their use by telephone companies. It has been user demand that has brought infrared light and now SONET-based infrared systems in place. Recently, the introduction of an unguided light introduced by Lucent Technologies operates at speeds up to 2.4 Gbps with a promised speed of up to 10 Gbps by end of 2000. This offers the connectivity to almost anyone who can afford the system, because the right of way is no longer an issue.

The Network Hierarchy (Post-1984)

After 1984, ownership of the network took a dramatic turn. AT&T separated itself from the Bell Operating Companies (BOCs), opening the door for more competition and new ventures. Equal access became a reality and users were no longer frustrated in their attempts to open their telecommunications networks to competition.

The Public-Switched Network

The U.S. public-switched network is the largest and the best in the world. Over the years, the network has penetrated to even the most remote locations around the country. The primary call-carrying capacity in the U.S. is done through the public-switched network. Because this is the environment AT&T and the BOCs built, we still refer to it as the Bell System. However, as we’ve already seen, significant changes have taken place to change that environment. The public network enables access to the end office, connects through the long-distance network, and delivers to the end. This makes the cycle complete. Many companies use the switched network exclusively, while others have created variations depending on need, finances, and size. The network is dynamic enough, however, to pass the call along longer routes through the hierarchy to complete the call in the first attempt wherever possible.

The North American Numbering Plan

The network-numbering plan was designed to enable a quick and discreet connection to any telephone in the country. The North American Numbering Plan , as it is called, works on a series of 10 numbers. As progress occurs, the use of Local Number Portability (LNP) and Intelligent Networks (IN) enables the competitors to break in and offer new services to the consumer. Note that there have been some changes in this numbering plan. When it originally was formulated, the telephone numbers were divided into three sets of sequences. The area codes were set to designate high-volume usage and enabled some number recognition tied to a state boundary. With the convergence in full swing, the numbering plan became a bottleneck. Now with the use of LNP, the Numbering Plan will completely become obsolete as we know it. No longer will we recognize the number by an area code and correlate it to a specific geographic area. LNP will make the number a fully portable entity. Moreover, 10-digit dialing in the age of convergence becomes the norm because of the multitude of area codes that will reside in a state.

Private Networks

Many companies created or built their own private networks in the past. These networks are usually cost-justified or based on the availability of lines, facilities, and special needs. Often these networks employ a mix of technologies, such as private microwaves, satellite communications, fiber optics, and infrared transmission. The convergence of the networks has further been deployed because of the mix of services that the telephone companies did not service well. Many companies with private networks have been subjected to criticisms because the networks were misunderstood. Often the networks were based on voice savings and could not be justified. Now that the telecommunications networks and systems are merging, the demand for higher-speed and more availability is driving either a private network or a hybrid.

Hybrid Networks

Some companies have to decide whether to use a private- or public-switched network for their voice, data, video, and Internet needs. Therefore, these organizations use a mix of services based on both private and public networks. The high-end usage is connected via private facilities creating a Virtual Private Network (VPN), while the lower-volume locations utilize the switched network. Installing private-line facilities comes from the integration of voice, data, video, graphics, and facsimile transmissions. Now VPNs are used on the Internet to guarantee speed, throughput, quality of service, and reliability. This new wave of VPN takes up where the voice VPNs left off. Only by combining these services across a common circuitry will many organizations realize a savings.

Equipment
Equipment in the telephony and telecommunications business is highly varied and complex. The mix of goods and services is as large as the human imagination, yet the standard types are the ones that constitute the ends on the network. The convergence and computerization of our equipment over the years has led to significant variations. The devices that hook up to the network are covered in various other chapters, but here is a summary of certain connections and their functions in the network:

• The Private branch exchange ( PBX)
• The modem (data communications device)
• The multiplexer (enables more users on a single line)
• Automatic call distributor (ACD)
• Voice mail system (VMS)
• Automated attendant (AA)
• Radio systems
• Cellular telephones
• Facsimile machines
• CATV connections
• Web-enabled call centers
• Integrated voice recognition and response systems

This is a sampling of the types of equipment and services you will encounter in dealing with Telecommunications Systems and convergence in this industry.

Telecommunications Concepts

noreply@blogger.com (Utsav Basu) — Sun, 17 May 2009 09:48:45 PDT

After a few ideas have sunk in, we move on to a high-speed data networking strategy, with the use of Frame Relay. After Frame Relay, we discuss the use of ATM for its merits and benefits. Next, we take the convergence a step farther and delve into the Frame and ATM inter networking applications—still a great way to carry our voice and data no matter how we slice and dice it.

Just when we thought it was safe to use these high-speed services across the WAN, we realize that local access is a problem. Entering into the discussion is the high-speed convergence in the local loop arena with the use of CATV and cable modems to access the Internet at LAN speeds. Mix in a little xDSL, and we start the fires burning on the local wires. The use of copper wires or cable TV is the hot issue in data access. From the discussion of the local loop, we then see the comparisons of a wireless local loop with LMDS and MMDS. These techniques are all based on a form of Microwave, so the comparison of microwave radio techniques is shown. Wireless portability is another hot area in the marketplace. Therefore, we compare and contrast the use of GSM, cellular, and personal communications’ services and capacities. Convergence is only as good as one’s ability to place the voice and data on the same links.

Leaving the low-end wireless services behind, we then enter into a discussion of the sky wave and satellite transmission for voice and data. No satellite transmission discussion would be worth anything without paying homage to the TCP and IP protocols on the satellite networks. Yet, the satellite services are now facing direct competition where the Low Earth Orbit satellite strategies are becoming ever popular. The use of Teledesic, Iridium, or Globalstar systems are merely transport systems. These pull the pieces together and will offer voice and data transmission for years to come. One could not go too far with the wireless-only world, so we back up and begin to
contrast the use of the wired world again. This time, we look at T1, T2, and T3 on copper or coax cable, which is a journey down memory lane for some. However, by adding a little fiber to the diet, we provide these digital architectures on SONET or SDH services. SONET makes the T1 and T3 look like fun! Topics include the ability to carry Frame Relay and ATM as the networks are now beginning to meld together. SONET is good, but if we use an older form of multiplexing (wavelength), we can get more yet from the fibers. So, we look at the benefits of dense-wave division multiplexing on the fiber to carry more SONET and more data.

With the infrastructure kicked around, the logical step is to complete this tour of the telecommunications arena with the introduction of the Internet, intranets, and extranets. Wow, this stuff really does come together! Using the Internet or the other two forms of nets, we can then carry our data transparently. What would convergence be without the voice? Therefore, the next step is to look at the use of voice over Internet Protocols (IPs).

Lastly, we have to come up with a management system to control all the pieces that we have grouped and bonded together. This is in the form of a simple network management protocol (SNMP) as the network management tool of choice. If all the converged pieces work, there is no issue. However, with all the variants discussed in this book, we must believe that “Murphy is alive and well!” Thus, all the pieces are formed together by groups, to form a homogenous network of Internets.

Basic Telecommunications Systems

When the FCC began removing regulatory barriers for the long distance and customer premises equipment (CPE) markets, its goal was to increase competition through the number of suppliers in these markets. Recently, consumers have begun to enjoy lower prices and new bundled service offerings. The local and long distance markets are examples of the new direction taken by the FCC in the 1980s to eliminate and mitigate the traditional telephone monopoly into a set of competitive markets. Although these two components of the monopoly have been stripped away, barriers still exist at the local access network—the portion of the public network that extends between the IEC network and the end user. The local loop and the basic telecommunications infrastructure are not as readily available as one would like to think. The growth of private network alternatives improves with facilities-based competition in the transport of communications services. The industry realizes that more than 500 competitive local exchange carriers have grown out of the deregulation of the monopolies. These CLECs include cable television networks, wireless telephone networks, local area networks (LANs), and metropolitan area networks. Incumbent local exchange carriers (ILECs) indicate that their networks are continually evolving into a multimedia platform capable of delivering a rich variety of text, imaging, and messaging services as a direct response to the competition. Many suggest that their networks are wide open, for all competitors. Imagine an open network —a network with well-defined interfaces accessible to all—allowing an unlimited number of entrants a means to offer competitive services limited only by their imagination and the capabilities of the local loop network facilities. If natural monopolies are still in the local exchange network, open access to these network resources must be fostered to promote a competitive market in spite of the monopolistic nature of the ILECs. The FCC continues to wrestle with how far it has to go and what requirements are necessary to open and equal access to the network. Network unbundling, the process of breaking the network into separate functional elements, opens the local access to competition. CLECs select unbundled components they need to provide their own service. If the unbundled price is still too expensive, the service provider will provide its own private resources. This is the facilities-based provider. All too often, we hear about new suppliers who offer high-speed services, better than the incumbent. Yet, these suppliers are typically using the Bell System’s wires to get to the consumer’s door. The only change that occurs is the person to whom we send the bill. Hardly a competitive local networking strategy. As a result, the new providers
(CATV, wireless local loop, IEC, and facilities-based CLEC) are now in the mode to provide their own facilities.

Components of the Telecommunications Networks

Telecommunications network components fall into logical or physical elements. A logical element is a software-defined network (SDN) or virtual private network (VPN) feature or capability. This SDN or VPN feature can be as simple as the number translation performed in a switch to establish a call. Switching systems have evolved into the use of external signaling systems to set up and tear down the call. These external physical and logical components formulate the basis of a network element. Moreover, Intelligent Networks (and Advanced Intelligent Networks) have surpassed the wildest expectations of the service provider. These logical extensions of the network bear higher revenue while opening the network up to a myriad of new services. Number portability can also be categorized into the logical elements because the number switching and logic are no longer bound to a specific system. A physical element is the actual switching element, such as the link or the matrices used internally. A network is made up of a unique sequence of logical elements implemented by physical elements. Given the local exchange network and local transport markets, open mandates had to be considered because the LEC has the power to stall competition. In many documented cases the LECs have purposefully dragged their feet to stall the competition and to discredit the new provider in the eyes of the customer. This is a matter of survival of the fittest. The ILECs have the edge over the network components because their networks were built over the past 120 + years. This is the basis for the deregulatory efforts in the networks, because the LECs are fighting to survive the onslaught of new providers who are in the cream-skimming mode. If access mandates are necessary, to what degree? These and other issues are driving the technological innovation, competition at the local
loop, and the development of higher capacity services in a very competitive manner.

The Local Loop

So much attention has been parlayed on the local loop. Nevertheless, is it a realistic expectation to use the network facilities for future high-speed services? Would the newer providers, such as the CATV companies, have an edge over the ILECs? These issues are the foundation of the network of the new millennium. The new providers will use whatever technology is available to attack the competition, including

• CATV
• Fiber-based architectures (FTTC, FTTH, HFC)
• Wireless microwave systems
• Wireless third-generation cellular systems
• Infrared and laser based wireless architectures
• Satellite and DSS type services
Regardless of the technology used, the demand never seems to be satisfied. Therefore, the field of competitors will continue to metamorphose as the demand dictates and as the revenues continue to attract new business.

The Movement Toward Fiber Optic Networks

A transmission link transports information from one location to another in a usable and understandable format. The three functional attributes of this link are-

1. Capacity
2. Condition
3. Quality of Service
The deregulation of the local exchange networks has led to significant improvements in one of the following criteria:
• Access to network capacity
• Access to intermediate points along the transmission path The transmission path may include pieces of the existing copper or newer fiber-based network architectures. The current copper-based loop limits opportunities.
• The transmission distances associated with the subscriber loop limit the amount of bandwidth available over twisted wire pair roughly to the DS1 rate of 1.5 Mbps. As broadband services become increasingly popular, the copper network severely constrains the broadband services.
• The current switched-star architecture runs at least one dedicated twisted pair from the central office to each customer’s door without any intermediate locations available to unbundle the transport segment. This precludes a lot of the innovation desired by the end user. Although the current copper-based network is unattractive to unbundle the physical transmission components, fiber-based networks offer many more opportunities. The local access network can be improved by telephone companies by deploying fiber in the future. The central office, nodes at remote sites and the curbside pedestal can all be improved with fiber-based architectures. These nodes serve as flexibility points where signals can be switched or multiplexed to the appropriate destination. A small percentage of lines are served by digital loop carrier (DLC) systems that incorporate a second flexibility point into the architecture at the remote node. The third flexibility point at the pedestal has been proposed for fiber-to-the-curb systems in the future. The bandwidth limitations of a fiber system are not due to the intrinsic properties of the
fiber, but the limitations of the switching, multiplexing, and transmission equipment connected to the fiber. This opens the world up for a myriad of new service offerings when fiber makes it to the consumer’s door. Third parties like Qwest and Level 3 are becoming the carrier’s carrier. They will install the fiber to the pedestal, the door, or to the backbone and sell the capacity to the ILEC or CLEC. This produces many attractive alternatives to the broadband networks for the future. No longer will bandwidth be the constraining factor; the application or the computer will be the bottleneck.

Because of the tremendous bandwidth available with fiber optic cable and the technological improvements in SONET and Dense Wave Division Multiplexing, virtually unlimited bandwidth will be available. This statement of course is contingent on the following caveats:

• The abundance of bandwidth is not likely to appear for some time.
• This bandwidth is available only over the fiber links. Yet, installation of new technology is a slow process. Fiber will be deployed in hybrid network architectures, which continue to utilize existing portions of the copper network. Consequently, until fiber is deployed all the way to the customer premises, portions of the network will continue to present the same speed and throughput limitations.

Digital Transfer Systems

The switching and multiplexing techniques characteristic of the transmission systems within the network are all digital. Currently, the network employs a synchronous transfer mode (STM) technique for switching and multiplexing these digital signals. The broadband networks of the future will continue to utilize a synchronous transmission hierarchy using the SONET standards defined by the ITU. SONET describes a family of broadband digital transport signals operating in 50 Mbps increments. As a result, wherever SONET equipment is used, the standard interfaces at the central office, remote nodes, or subscriber premises will be multiples of these rates. Above the physical layer, however, changes are now underway that move away from the
synchronous communications modes. The asynchronous transfer mode (ATM) is the preferred method of transporting at the data link layer. ATM uses the best of packet switching and routing techniques to carry information signals, regardless of the desired bandwidth, over one high-speed switching fabric. Using fixed-length cells, the information is processed at higher speeds, reducing some of the original latency in the network. These cells then combine with the cells of other signals across a single highspeed channel like a SONET OC − 48. In time division multiplexing (TDM), timing is crucial. In ATM, timing is statistically multiplexed (STDM) so the timing is less crucial at the data link layer. The cells fit into the payload of the SONET frame structure for transmission where the timing is again used by the physical layer devices. ATM will use a combined switching and multiplexing service at the cell level. Continued use of SONET multiplexers will combine and separate SONET signals carrying ATM cells. What distinguishes ATM from a synchronous approach is that subscribers have the ability to customize their use of the bandwidth without being constrained to the channel data rates.

When the intelligent networks are fully implemented, the logical network components will be separated from the physical switching element—where the physical component of a current digital switch consists of 64 Kbps (DS0) access to the network switch. ATM should improve the capability to separate the physical switching elements of the network. The attributes of the ATM switch, which could facilitate more modularity, is the bandwidth flexibility. Because each information signal is segmented into cells, switching is performed in much smaller increments. Current digital switching elements switch a DS0 signal whether the full bandwidth is needed or not. With ATM, the switching element resources can be much more efficiently matched to the bandwidth requirements of the user. Access to the ATM switch will be specified according to the maximum data rate forecasted for the particular access arrangement, instead of specifying the number of DS0 circuits required, as is the case today with digital switches.

The Intelligent Networks of Tomorrow

The ILECs have been developing the AIN to provide new services or to customize current services based on the user demand. The Central Office switches contain the necessary software to facilitate these enhanced features. The manufacturers of the systems have fully embodied their application software with the operating systems software within the switch to create a simple interface for the carriers. When new features are added, the integrated software must be fully tested by the switch manufacturer. The limitations of a centralized architecture caused the vendors and manufacturers concern. Now, as intelligent services are deployed, the movement is to a distributed architecture and Intelligent Peripheral devices on the network. The LECs use a network architecture, which enables efficient and rapid network deployment. The single most important feature of AIN is its flexibility to configure the network according to the characteristics of the service. The modular architecture allows the addition of adjunct processors, such as voice processing equipment, data communication gateways, video services, and directory look-up features to the network without major modifications. These peripheral devices (servers) provide local customer database information and act like the intelligent centralized architectures of old. The basic architecture of the AIN takes these application functions and breaks them into a
collection of functionally specific components. Ultimately, AIN allows modifications to application software without having to alter the operating system of the switch.

Summary

The telecommunications systems include the variations of the local loop and the changes taking place within that first (or last) mile. As the migration moves away from the local copper-based cable plant (a slow evolution for sure), the movement will be to other forms of communications subsystems to include the use of

• Fiber optics
• Coax cable
• Radio-based systems
• Light-based systems
• Hybrids of the preceding

These changes will take users and carriers alike into the new millennium. Using the CATV modem technologies on coax, the fiber-based SONET architectures in the backbone (and ultimately in the local loop), and copper wires in the xDSL technologies all combine to bring higher speed access. After access is accomplished, the use of the SONET-based protocols and multiplexing systems creates an environment for the orchestration of newer services and features that will be bandwidth intensive. The SONET systems will be used to step up to the
challenges of the 2000s. ATM will add a new dimension to the access methods and the transport of the broadband information through the use of STDM and cell-based transmission. No longer will the network suppliers have to commit specific fixed bandwidth to an application that only rarely uses the service. Instead, the services will merely use the cells as necessary to perform the functionality needed. Wireless local loop services are relatively new in the broadband arena but will play a significant role in the future. The untethered ability to access the network no matter where you are will be attractive to a large new population of users. Access to low-speed
voice and data services are achievable today. However, the demand for real-time voice, data, video, and multimedia applications from a portable device is what the new generation of networks must accommodate. The broadband convergence will set the stage for all future development. Today speeds are set up in the kilobits to megabits per second range. The broadband networks of the future will have to deal with demands for multi-megabit speeds up to the gigabit per second speeds. Through each interface, the carriers must be able to preserve
as much of their infrastructure as possible so that forklift technological changes are not forced upon them. The business case for the evolution of the broadband convergence is one that mimics a classical business model. Using a 7-15 year return on investment model, the carriers must see the benefit of profitability before they install the architectural changes demanded today.

Frame Relay Layers

noreply@blogger.com (Utsav Basu) — Wed, 29 Apr 2009 21:51:27 PDT

Frame Relay Layers

Frame Relay has only physical and data link layers.

Physical Layer

No specific protocol is defined for the physical layer in Frame Relay. Instead, it is left to the implementer to use whatever is available. Frame Relay supports any of the protocols recognized by ANSI.

Data Link Layer

At the data link layer, Frame Relay uses a simple protocol that does not support flow or error control. It only has an error detection mechanism.

##Address (DLCI) field. The first 6 bits of the first byte makes up the first part of the DLCI. The second part of the DLCI uses the first 4 bits of the second byte. These bits are part of the 10-bit data link connection identifier defined by the standard. We will discuss extended addressing at the end of this section.

##Command/response (C/R). The command/response (C/R) bit is provided to allow upper layers to identify a frame as either a command or a response. It is not used by the Frame Relay protocol.

##Extended address (EA). The extended address (EA) bit indicates whether the current byte is the final byte of the address. An EA of 0 means that another address byte is to follow (extended addressing is discussed later). An EA of 1 means that the current byte is the final one.

##Forward explicit congestion notification (FECN). The forward explicit congestion notification (FECN) bit can be set by any switch to indicate that traffic is congested. This bit informs the destination that congestion has occurred. In this way, the destination knows that it should expect delay or a loss of packets.

##Backward explicit congestion notification (BECN). The backward explicit congestion notification (BECN) bit is set (in frames that travel in the other direction) to indicate a congestion problem in the network. This bit informs the sender that con- gestion has occurred. In this way, the source knows it needs to slow down to prevent the loss of packets.

##Discard eligibility (DE). The discard eligibility (DE) bit indicates the priority level of the frame. In emergency situations, switches may have to discard frames to relieve bottlenecks and keep the network from collapsing due to overload. When set (DE 1), this bit tells the network to discard this frame if there is congestion. This bit can be set either by the sender of the frames (user) or by any switch in the network.

Extended Address

To increase the range of DLCIs, the Frame Relay address has been extended from the original 2-byte address to 3- or 4-byte addresses. Figure 18.4 shows the different addresses. Note that the EA field defines the number of bytes; it is 1 in the last byte of the addres, and it is 0 in the other bytes. Note that in the 3- and 4-byte formats, the bit before the last bit is set to 0.

FRADs

To handle frames arriving from other protocols, Frame Relay uses a device called a Frame Relay assembler/disassembler (FRAD). A FRAD assembles and disassembles frames coming from other protocols to allow them to be carried by Frame Relay frames. A FRAD can be implemented as a separate device or as part of a switch.

VOFR

Frame Relay networks offer an option called Voice Over Frame Relay (VOFR) that sends voice through the network. Voice is digitized using PCM and then compressed. The result is sent as data frames over the network. This feature allows the inexpensive sending of voice over long distances. However, note that the quality of voice is not as good as voice over a circuit-switched network such as the telephone network. Also, the varying delay mentioned earlier sometimes corrupts real-time voice.

LMI

Frame Relay was originally designed to provide PVC connections. There was not, therefore, a provision for controlling or managing interfaces. Local Management Information (LMI) is a protocol added recently to the Frame Relay protocol to provide more management features. In particular, LMI can provide -----

1. keep-alive mechanism to check if data are flowing.

2. multicast mechanism to allow a local end system to send frames to more than one remote end system.

3. mechanism to allow an end system to check the status of a switch (e.g., to see if the switch is congested).

Virtual-Circuit Networks

noreply@blogger.com (Utsav Basu) — Wed, 29 Apr 2009 21:47:55 PDT

Virtual-Circuit Networks.' Frame Relay and ATM

In previous post , we discussed switching techniques. We said that there are three types of switching: circuit switching, packet switching, and message switching. We also mentioned that packet switching can use two approaches: the virtual-circuit approach and the datagram approach.

we show how the virtual-circuit approach can be used in wide-area networks. Two common WAN technologies use virtual-circuit switching. Frame Relay is a relatively high-speed protocol that can provide some services not available in other WAN technologies such as DSL, cable TV, and T lines. ATM, as a high-speed protocol, can be the superhighway of communication when it deploys physical layer carriers such as SONET.

We first discuss Frame Relay. We then discuss ATM in greater detail. Finally, we show how ATM technology, which was originally designed as a WAN technology, can also be used in LAN technology, ATM LANs.

FRAME RELAY

Frame Relay is a virtual-circuit wide-area network that was designed in response to demands for a new type of WAN in the late 1980s and early 1990s.

1. Prior to Frame Relay, some organizations were using a virtual-circuit switching network called X.25 that performed switching at the network layer. For example, the Intemet, which needs wide-area networks to carry its packets from one place to another, used X.25. And X.25 is still being used by the Internet, but it is being replaced by other WANs. However, X.25 has several drawbacks:

a. X.25 has a low 64-kbps data rate. By the 1990s, there was a need for higher data-rate WANs.

b. X.25 has extensive flow and error control at both the data link layer and the network layer. This was so because X.25 was designed in the 1970s, when the available transmission media were more prone to errors. Flow and error control at both layers create a large overhead and slow down transmissions. X.25 requires acknowledgments for both data link layer frames and network layer packets that are sent between nodes and between source and destination.

c. Originally X.25 was designed for private use, not for the Internet. X.25 has its own network layer. This means that the user's data are encapsulated in the network layer packets of X.25. The Internet, however, has its own network layer, which means if the Internet wants to use X.25, the Internet must deliver its network layer packet, called a datagram, to X.25 for encapsulation in the X.25 packet. This doubles the overhead.

2. Disappointed with X.25, some organizations started their own private WAN by leasing T- 1 or T-3 lines from public service providers. This approach also has some drawbacks.

a. If an organization has n branches spread over an area, it needs n(n - 1)/2 T- 1 or T-3 lines. The organization pays for all these lines although it may use the lines only 10 percent of the time. This can be very costly:

b. The services provided by T-1 and T-3 lines assume that the user has fixed-rate data all the time. For example, a T-1 line is designed for a user who wants to use the line at a consistent 1.544 Mbps. This type of service is not suitable for the many users today that need to send bursty data. For example, a user may want to send data at 6 Mbps for 2 s, 0 Mbps (nothing) for 7 s, and 3.44 Mbps for 1 s for a total of 15.44 Mbits during a period of 10 s. Although the average
data rate is still 1.544 Mbps, the T-1 line cannot accept this type of demand because it is designed for fixed-rate data, not bursty data. Bursty data require what is called bandwidth on demand. The user needs different bandwidth allocations at different times. In response to the above drawbacks, Frame Relay was designed. Frame Relay is a wide area network with the following features:

1. Frame Relay operates at a higher speed (1.544 Mbps and recently 44.376 Mbps). This means that it can easily be used instead of a mesh ofT-1 or T-3 lines.

2. Frame Relay operates in just the physical and data link layers. This means it can easily be used as a backbone network to provide services to protocols that already have a network layer protocol, such as the Internet.

3. Frame Relay allows bursty data.

4. Frame Relay allows a frame size of 9000 bytes, which can accommodate all local area network frame sizes.

5. Frame Relay is less expensive than other traditional WANs.

6. Frame Relay has error detection at the data link layer only. There is no flow control or error control. There is not even a retransmission policy if a frame is damaged; it is silently dropped. Frame Relay was designed in this way to provide fast transmission capability for more reliable media and for those protocols that have flow and error control at the higher layers.

Architecture

Frame Relay provides permanent virtual circuits and switched virtual circuits. The routers are
used,to connect LANs and WANs in the Internet. In the figure, the Frame Relay WAN is used as one link in the global Internet.
Virtual Circuits

Frame Relay is a virtual circuit network. A virtual circuit in Frame Relay is identified by a number called a data link connection identifier (DLCI).

Permanent Versus Switched Virtual Circuits

A source and a destination may choose to have a permanent virtual circuit (PVC). In this case, the connection setup is simple. The corresponding table entry is recorded for all switches by the administrator (remotely and electronically, of course). An outgoing DLCI is given to the source, and an incoming DLCI is given to the destination. PVC connections have two drawbacks. First, they are costly because two parties pay for the connection all the time even when it is not in use. Second, a connection is created from one source to one single destination. If a source needs connections with several destinations, it needs a PVC for each connection. An alternate approach is the switched virtual circuit (SVC). The SVC creates a temporary, short connection that exists only when data are being transferred between source and destination. An SVC requires establishing and terminating phases.

Switches

Each switch in a Frame Relay network has a table to route frames. The table matches an incoming port-DLCI combination with an outgoing port-DLCI combination. The only difference is that VCIs are replaced by DLCIs.

SONET NETWORKS

noreply@blogger.com (Utsav Basu) — Mon, 27 Jul 2009 20:55:32 PDT

SONET NETWORKS

Using SONET equipment, we can create a SONET network that can be used as a high-speed backbone carrying loads from other networks such as ATM (Chapter 18) or IP (Chapter 20). We can roughly divide SOlNET networks into three categories: linear, ting, and mesh networks.

Linear Networks

A linear SONET network can be point-to-point or multipoint. Point-to-Point Network. A point-to-point network is normally made of an STS multiplexer, an STS demultiplexer, and zero or more regenerators with no add/drop multiplexers, as shown in Figure 17.18. The signal flow can be unidirectional or bidirectional,

Multipoint Network

A multipoint network uses ADMs to allow the communications between several terminals. An ADM removes the signal belonging to the terminal connected to it and adds the signal transmitted from another terminal. Each terminal can send data to one or more downstream terminals. which each terminal can send data only to the downstream terminals, but the a multipoint network can be bidirectional, too.

Automatic Protection Switching

To create protection against failure in linear networks, SONET defines automatic protection switching (APS). APS in linear networks is defined at the line layer, which means the protection is between two ADMs or a pair of STS multiplexer/multiplexers. The idea is to provide redundancy; a redundant line (fiber) can be used in case of failure in the main one. The main line is referred to as the work line and the redundant line as the protection line. Three schemes are common for protection in linear channels:
one-plus-one, one-to-one, and one-to-many.

One-Plus-One APS In this scheme, there are normally two lines: one working line and one protection line. Both lines are active all the time. The sending multiplexer

sends the same data on both lines; the receiver multiplexer monitors the line and chooses the one with the better quality. If one of the lines fails, it loses its signal, and, of course, the other line is selected at the receiver. Although, the failure recovery for this scheme is instantaneous, the scheme is inefficient because two times the bandwidth is required. Note that one-plus-one switching is done at the path layer.

One-to-One APS In this scheme, which looks like the one-plus-one scheme, there is also one working line and one protection line. However, the data are normally sent on the working line until it fails. At this time, the receiver, using the reverse channel, informs the sender to use the protection line instead. Obviously, the failure recovery is slower than that of the one-plus-scheme, but this scheme is more efficient because the protection line can be used for data transfer when it is not used to replace the working line. Note that the one-to-one switching is done at the line layer.

One-to-Many APS This scheme is similar to the one-to-one scheme except that there is only one protection line for many working lines. When a failure occurs in one of the working lines, the protection line takes control until the failed line is repaired. It is not as secure as the one-to-one scheme because if more than one working line fails at the same time, the protection line can replace only one of them. Note that one-to-many APS is done at the line layer.

Ring Networks

ADMs make it possible to have SONET ring networks. SONET rings can be used in either a unidirectional or a bidirectional configuration. In each case, we can add extra rings to make the network self-healing, capable of self-recovery from line failure. Unidirectional Path Switching Ring

A unidirectional path switching ring (UPSR) is a unidirectional network with two rings: one ring used as the working ring and the other as the protection ring. The idea is similar to the one-plus-one APS scheme we discussed in a linear network. The same signal flows through both rings, one clockwise and the other counterclockwise. It is called UPSR because monitoring is done at the path layer. A node receives two copies of the electrical signals at the path layer, compares them, and chooses the one with the better quality. If part of a ring between two ADMs fails, the other ring still can guarantee the continuation of data flow. UPSR, like the one-plus-one scheme, has fast failure recovery, but it is not efficient because we need to have two rings that do the job of one. Half of the bandwidth is wasted.

Although we have chosen one sender and three receivers in the figure, there can be many other configurations. The sender uses a two-way connection to send data to both rings simultaneously; the receiver uses selecting switches to select the ring with better signal quality. We have used one STS multiplexer and three STS alemultiplexers to emphasize that nodes operate on the path layer.

Bidirectional Line Switching Ring

Another alternative in a SONET ring network is bidirectional line switching ring (BLSR). In this case, communication is bidirectional, which means that we need two rings for working lines. We also need two rings for protection lines. This means BLSR uses four rings. The operation, however, is similar to the one-to-one APS scheme. If a working ring in one direction between two nodes fails, the receiving node can use the reverse ring to inform the upstream node in the failed direction to use the protection ring. The network can recover in several different failure situations that we do not discuss here. Note that the discovery of a failure in BLSR is at the line layer, not the path layer. The ADMs find the failure and inform the adjacent nodes to use the protection rings.

Combination of Rings

SONET networks today use a combination of interconnected rings to create services in a wide area. For example, a SONET network may have a regional ring, several local rings, and many site rings to give services to a wide area. These rings can be UPSR, BLSR, or a combination of both.
Mesh Networks

One problem with ring networks is the lack of scalability. When the traffic in a ring increases, we need to upgrade not only the lines, but also the ADMs. In this situation, a mesh network with switches probably give better performance. A switch in a network mesh is called a cross-connect. A cross-connect, like other switches we have seen, has input and output ports. In an input port, the switch takes an OC-n signal, changes it to an STS-n signal, demultiplexes it into the corresponding STS-1 signals, and sends each STS-1 signal to the appropriate output port. An output port takes STS-1 signals coming from different input ports, multiplexes them into an STS-n signal, and makes an OC-n signal for transmission.

VIRTUAL TRIBUTARIES

SONET is designed to carry broadband payloads. Current digital hierarchy data rates (DS-1 to DS~3), however, are lower than STS-1. To make SONET backward-compatible with the current hierarchy, its frame design includes a system of virtual tributaries (VTs) . A virtual tributary is a partial payload that can be inserted into an STS-1 and combined with other partial payloads to fill out the frame. Instead of using all 86 payload columns of an STS-1 frame for data from one source, we can sub- divide the SPE and call each component a VT.

Types of VTs

Four types of VTs have been defined to accommodate existing digital hierarchies Notice that the number of columns allowed for each type of VT can be determined by doubling the type identification number

(VT1.5 gets three columns, VT2 gets four columns, etc.).
VT1.5 accommodates the U.S. DS-1 service (1.544 Mbps).
VT2 accommodates the European CEPT-1 service (2.048 Mbps).
VT3 accommodates the DS-1C service (fractional DS-l, 3.152 Mbps).
VT6 accommodates the DS-2 service (6.312 Mbps).

When two or more tributaries are inserted into a single STS-1 frame, they are interleaved column by column. SONET provides mechanisms for identifying each VT and separating them without demultiplexing the entire stream.

Encapsulation

noreply@blogger.com (Utsav Basu) — Wed, 29 Apr 2009 21:37:06 PDT

Encapsulation

The previous discussion reveals that an SPE needs to be encapsulated in an STS-1 frame. Encapsulation may create two problems that are handled elegantly by SONET using pointers (H1 to H3). We discuss the use of these bytes in this section.

Offsetting

SONET allows one SPE to span two frames, part of the SPE is in the first frame and part is in the second. This may happen when one SPE that is to be encapsulated is not aligned time-wise with the passing synchronized frames. SPE bytes are divided between the two frames. The first set of bytes is encapsulated in the first frame; the second set is encapsulated in the second frame. The figure also shows the path overhead, which is aligned with the section/line overhead of any frame. The question is, How does the SONET multiplexer know where the SPE starts or ends in the frame? The solution is the use of pointers H1 and H2 to define the beginning of the SPE; the end can be found because each SPE has a fixed number of bytes. SONET allows the offsetting of an SPE with respect to an STS-1 frame. To find the beginning of each SPE in a frame, we need two pointers H1 and H2 in the line overhead. Note that these pointers are located in the line overhead because the encapsulation occurs at a multiplexer.

the beginning of the SPEs. Note that we need 2 bytes to define the position of a byte in a frame; a frame has 810 bytes, which cannot be defined using 1 byte.

STS MULTIPLEXING

In SONET, frames of lower rate can be synchronously time-division multiplexed into a higher-rate frame. For example, three STS-1 signals (channels) can be combined into one STS-3 signal (channel), four STS-3s can be multiplexed into one STS-12.

Multiplexing is synchronous TDM, and all clocks in the network are locked to a master clock to achieve synchronization.

We need to mention that multiplexing can also take place at the higher data rates. For example, four STS-3 signals can be multiplexed into an STS-12 signal. However, the STS-3 signals need to first be demultiplexed into 12 STS-1 signals, and then these twelve signals need to be multiplexed into an STS-12 signal. The reason for this extra work will be clear after our discussion on byte interleaving.

Byte Interleaving

Synchronous TDM multiplexing in SONET is achieved by using byte interleaving. For example, when three STS-1 signals are multliplexed into one STS-3 signal, each set of 3 bytes in the STS-3 signal is associated with 1 byte from each STS- 1 signal.

Concatenated Signal

In normal operation of the SONET, an STS-n signal is made of n multiplexed STS-1 signals. Sometimes, we have a signal with a data rate higher than what an STS- 1 can carry. In this case, SONET allows us to create an STS-n signal which is not considered as n STS-1 signals; it is one STS-n signal (channel) that cannot be demultiplexed into n STS- 1 signals. To specify that the signal cannot be demultiplexed, the suffix c (for concatenated) is added to the name of the signal. For example, STS-3c is a signal that cannot be demultiplexed into three STS-1 signals. However, we need to know that the whole payload in an STS-3c signal is one SPE, which means that we have only one column (9 bytes) of path overhead. The used data in this case occupy 260 columns,

Add/Drop Multiplexer

Multiplexing of several STS-1 signals into an STS-n signal is done at the STS multiplexer (at the path layer). Demultiplexing of an STS-n signal into STS- 1 components is done at the STS demultiplexer. In between, however, SONET uses add/drop multiplexers that can replace a signal with another one. We need to know that this is not demultiplexing/multiplexing in the conventional sense. An add/drop multiplexer operates at the line layer. An add/drop multiplexer does not create section, line, or path overhead. It almost acts as a switch; it removes one STS-1 signal and adds another one. The type of signal at the input and output of an add/drop multiplexer is the same (both STS-3 or both STS-12, for example). The add/drop multiplexer (ADM) only removes the corresponding bytes and replaces them with the new bytes
(including the bytes in the section and line overhead).

STS-l frame: line overhead

noreply@blogger.com (Utsav Basu) — Wed, 29 Apr 2009 21:33:52 PDT

STS-l frame: line overhead

Line parity byte (B2). Byte B2 is for bit interleaved parity. It is for error checking of the frame over a line (between two multiplexers). In an STS-n frame, B2 is calculated for all bytes in the previous STS-1 frame and inserted at the B2 byte for that frame. In other words, in a STS-3 frame, there are three B2 bytes, each calculated for one STS-1 frame. Contrast this byte with B 1 in the section overhead.

Data communication channel bytes (D4 to D12). The line overhead D bytes (D4 to D12) in consecutive frames form a 576-kbps channel that provides the same service as the D l-D3 bytes (OA&M), but at the line rather than the section level (between multiplexers).

Order wire byte (E2). The E2 bytes in consecutive frames form a 64-kbps channel that provides the same functions as the E1 order wire byte, but at the line level.

Pointer bytes (HI, H2, and H3). Bytes H1, H2, and H3 are pointers. The first two bytes are used to show the offset of the SPE in the frame; the third is used for justification. We show the use of these bytes later.

Automatic protection switching bytes (K1 and K2). The K1 and K2 bytes in consecutive frames form a 128-kbps channel used for automatic detection of problems in line-terminating equipment.

Growth bytes (Z1 and Z2). The Z1 and Z2 bytes are reserved for future use.

Synchronous Payload Envelope

The synchronous payload envelope (SPE) contains the user data and the overhead related to the user data (path overhead). One SPE does not necessarily fit it into one STS- 1 frame; it may be split between two frames, as we will see shortly. This means that the path overhead, the leftmost column of an SPE, does not necessarily align with the section or line overhead. The path overhead must be added first to the user data to create an SPE, and then an SPE can be inserted into one or two frames. Path overhead consists of 9 bytes.

Path parity byte (B3). Byte B3 is for bit interleaved parity, like bytes B1 and B2, but calculated over SPE bits. It is actually calculated over the previous SPE in the stream.

Path signal label byte (C2). Byte C2 is the path identification byte. It is used to identify different protocols used at higher levels (such as IP or ATM) whose data are being carried in the SPE.

Path user channel byte (F2). The F2 bytes in consecutive frames, like the F1 bytes, form a 64-kbps channel that is reserved for user needs, but at the path level.

Path status byte (G1). Byte G1 is sent by the receiver to communicate its status to the sender. It is sent on the reverse channel when the communication is duplex. We will see its use in the linear or ring networks later in the chapten

Multiframe indicator (H4). Byte H4 is the multiframe indicator. It indicates payloads that cannot fit into a single frame. For example, virtual tributaries can be combined to form a frame that is larger than an SPE frame and need to be divided into different frames. Virtual tributaries are discussed in the next section. Path trace byte (J1). The J1 bytes in consecutive frames form a 64-kbps channel used for tracking the path. The J1 byte sends a continuous 64-byte string to verify the connection. The choice of the string is left to the application program. The receiver compares each pattern with the previous one to ensure nothing is wrong with the communication at the path layer.

Growth bytes (Z3, Z4, and Z5). Bytes Z3, Z4, and Z5 are reserved for future use.

SONET LAYERS

noreply@blogger.com (Utsav Basu) — Wed, 29 Apr 2009 21:31:12 PDT

SONET LAYERS

The SONET standard includes four functional layers: the photonic, the section, the line, and the path layer. They correspond to both the physical and the data link layers . The headers added to the frame at the various layers are discussed later in this chapter.

Path Layer

The path layer is responsible for the movement of a signal from its optical source to its optical destination. At the optical source, the signal is changed from an electronic form into an optical form, multiplexed with other signals, and encapsulated in a frame. At the optical destination, the received frame is demultiplexed, and the individual optical signals are changed back into their electronic forms. Path layer overhead is added at this layer. STS multiplexers provide path layer functions.

Line Layer

The line layer is responsible for the movement of a signal across a physical line. Line layer overhead is added to the frame at this layer. STS multiplexers and add/drop multiplexers provide line layer functions.

Section Layer

The section layer is responsible for the movement of a signal across a physical section. It handles framing, scrambling, and error control. Section layer overhead is added to the frame at this layer.

Photonic Layer

The photonic layer corresponds to the physical layer of the OSI model. It includes physical specifications for the optical fiber channel, the sensitivity of the receiver, multiplexing functions, and so on. SONET uses NRZ encoding with the presence of light representing 1 and the absence of light representing 0.

Device-Layer Relationships

an STS multiplexer is a four-layer device. An add/drop multiplexer is a three-layer device. A regenerator is a two-layer device.

SONET FRAMES

Each synchronous transfer signal STS-n is composed of 8000 frames. Each frame is a two-dimensional matrix of bytes with 9 rows by 90 x n columns. For example, STS- 1 frame is 9 rows by 90 columns (810 bytes), and an STS-3 is 9 rows by 270 columns (2430 bytes).

Frame, Byte, and Bit Transmission

One of the interesting points about SONET is that each STS-n signal is transmitted at a fixed rate of 8000 frames per second. This is the rate at which voice is digitized . For each frame the bytes are transmitted from the left to the right, top to the bottom. For each byte, the bits are transmitted from the most significant to the least significant (left to right).

If we sample a voice signal and use 8 bits (1 byte) for each sample, we can say that each byte in a SONET frame can carry information from a digitized voice channel. In other words, an STS-1 signal can carry 774 voice channels simultaneously (810 minus required bytes for overhead).

STS-1 Frame Format

SONET frame is a matrix of 9 rows of 90 bytes (octets) each, for a total of 810 bytes. The first three columns of the frame are used for section and line overhead. The upper three rows of the first three columns are used for section overhead (SOH). The lower six are line overhead (LOH). The rest of the frame is called the synchronous payload envelope (SPE). It contains user data and path overhead (POH) needed at the user data level. We will discuss the format of the SPE shortly.

Section Overhead

Alignment bytes (A1 and A2). Bytes A1 and A2 are used for framing and synchronization and are called alignment bytes. These bytes alert a receiver that a frame is arriving and give the receiver a predetermined bit pattern on which to syn- chronize. The bit patterns for these two bytes in hexadecimal are 0xF628. The bytes serve as a flag.

Section parity byte (B1). Byte B1 is for bit interleaved parity (BIP-8). Its value is calculated over all bytes of the previous frame. In other words, the ith bit of this byte is the parity bit calculated over all ith bits of the previous STS-n frame. The value of this byte is filled only for the first STS-1 in an STS-n frame. In other words, although an STS-n frame has n B 1 bytes, as we will see later, only the first byte has this value; the rest are filled with Os.

Identification byte (C1). Byte C1 carries the identity of the STS-1 frame. This byte is necessary when multiple STS-ls are multiplexed to create a higher-rate STS (STS-3, STS-9, STS-12, etc.). Information in this byte allows the various signals to be recog- nized easily upon demultiplexing. For example, in an STS-3 signal, the value of the C 1 byte is 1 for the first STS- 1; it is 2 for the second; and it is 3 for the third.

Management bytes (D1, D2, and D3). Bytes D1, D2, and D3 together form a 192-kbps channel (3 x 8000 x 8) called the data communication channel. This chan- nel is required for operation, administration, and maintenance (OA&M) signaling.

Order wire byte (El). Byte E1 is the order wire byte. Order wire bytes in consecutive frames form a channel of 64 kbps (8000 frames per second times 8 bits per

frame). This channel is used for communication between regenerators, or between terminals and regenerators.

User's byte (F1). The F1 bytes in consecutive frames form a 64-kbps channel that is reserved for user needs at the section level.

SONET/SDH

noreply@blogger.com (Utsav Basu) — Wed, 29 Apr 2009 21:27:45 PDT

SONET/SDH

SONET, that is used as a transport network to carry loads from other WANs. We first discuss SONET as a protocol, and we then show how SONET networks can be constructed from the standards defined in the protocol. The high bandwidths of fiber-optic cable are suitable for today's high-data-rate technologies (such as video conferencing) and for carrying large numbers of lower-rate technologies at the same time. For this reason, the importance of fiber optics grows in conjunction with the development of technologies requiring high data rates or wide bandwidths for transmission. With their prominence came a need for standardization. The United States (ANSI) and Europe (ITU-T) have responded by defining standards that, though independent, are fundamentally similar and ultimately compatible. The ANSI standard is called the Synchronous Optical Network (SONET). The ITU-T standard is called the Synchronous Digital Hierarchy (SDH).

ARCHITECTURE

Let us first introduce the architecture of a SONET system: signals, devices, and connections. Signals SONET defines a hierarchy of electrical signaling levels called synchronous transport signals (STSs). Each STS level (STS-1 to STS-192) supports a certain data rate, specified in megabits per second . The corresponding optical signals are called optical carriers (OCs). SDH specifies a similar system called a synchronous transport module (STM). STM is intended to be compatible with existing European hierarchies, such as E lines, and with STS levels. To this end, the lowest STM level, STM-1, is defined as 155.520 Mbps, which is exactly equal to STS-3.

SONET Devices

SONET transmission relies on three basic devices: STS multiplexers/demultiplexers, regenerators, add/drop multiplexers and terminals.

STS Multtiplexer/Detnultiplexer

STS multiplexers/demultiplexers mark the beginning points and endpoints of a SONET link. They provide the interface between an electrical tributary network and the optical network. An STS multiplexer multiplexes signals from multiple electrical sources and creates the corresponding OC signal. An STS demultiplexer demultiplexes an optical OC signal into corresponding electric signals.

Regenerator

Regenerators extend the length of the links. A regenerator is a repeater that takes a received optical signal (OC-n), demodulates it into the corresponding electric signal (STS-n), regenerates the electric signal, and finally modulates the electric signal into its correspondent OC-n signal. A SONET regenerator replaces some of the existing overhead information (header information) with new information.

Add/drop Multiplexer

Add/drop multiplexers allow insertion and extraction of signals. An add/drop multiplexer (ADM) can add STSs coming from different sources into a given path or can remove a desired signal from a path and redirect it without demultiplexing the entire signal. Instead of relying on timing and bit positions, add/drop multiplexers use header information such as addresses and pointers (described later in this section) to identify individual streams.

In the simple configuration , a number of incoming electronic signals are fed into an STS multiplexer, where they are combined into a single optical signal. The optical signal is transmitted to a regenerator, where it is recreated without the noise it has picked up in transit. The regenerated signals from a number of sources are then fed into an add/drop multiplexer. The add/drop multiplexer reorganizes these signals, if necessary, and sends them out as directed by information in the data frames. These remultiplexed signals are sent to another regenerator and from there to the receiving STS demultiplexer, where they are returned to a format usable by the receiving links.

Terminals

A terminal is a device that uses the services of a SONET network. For example, in the Internet, a terminal can be a router that needs to send packets to another router at the other side of a SONET network.
Connections

The devices defined in the previous section are connected using sections, lines, and paths.

Sections

A section is the optical link connecting two neighbor devices: multiplexer to multiplexer, multiplexer to regenerator, or regenerator to regenerator.

Lines

A line is the portion of the network between two multiplexers: STS multiplexer to add/ drop multiplexer, two add/drop multiplexers, or two STS multiplexers.

Paths

A path is the end-to-end portion of the network between two STS multiplexers. In a simple SONET of two STS multiplexers linked directly to each other, the section, line, and path are the same.

GSM

noreply@blogger.com (Utsav Basu) — Mon, 27 Apr 2009 23:05:40 PDT

GSM

The Global System for Mobile Communication (GSM) is a European standard that was developed to provide a common second-generation technology for all Europe. The aim was to replace a number of incompatible first-generation technologies. Bands GSM uses two bands for duplex communication. Each band is 25 MHz in width, shifted toward 900 MHz, Each band is divided into 124 channels of 200 kHz separated by guard bands.

Each voice channel is digitized and compressed to a 13-kbps digital signal. Each slot carries 156.25 bits. Eight slots share a frame (TDMA). Twenty-six frames also share a multiframe (TDMA). We can calculate the bit rate of each channel as follows:

Channel datarat = (i/120 ms) x 26 X 8 X 156.25 = 270.8 kbps

Each 270.8-kbps digital channel modulates a carrier using GMSK (a form of FSK used mainly in European systems); the result is a 200-kHz analog signal. Finally 124 analog channels of 200 kHz are combined using FDMA. The result is a 25-MHz band. Figure 16.9 shows the user data and overhead in a multiframe. The reader may have noticed the large amount of overhead in TDMA. The user data are only 65 bits per slot. The system adds extra bits for error correction to make it 114 bits per slot. To this, control bits are added to bring it up to 156.25 bits per slot. Eight slots are encapsulated in a frame. Twenty-four traffic frames and two additional control frames make a multiframe. A multiframe has a duration of 120 ms. However, the architecture does define superframes and hyperframes that do not add any overhead; we will not discuss them here.

Reuse Factor Because of the complex error correction mechanism, GSM allows a reuse factor as low as 3.

IS-95

One of the dominant second-generation standards in North America is Interim Standard 95 (IS-95). It is based on CDMA and DSSS. Bands and Channels IS-95 uses two bands for duplex communication. The bands can be the traditional ISM 800-MHz band or the ISM 1900-MHz band. Each band is divided into 20 channels of 1.228 MHz separated by guard bands. Each service provider is allotted 10 channels. IS-95 can be used in parallel with AMPS. Each IS-95 channel is equivalent to 41 AMPS channels (41 x 30 kHz = 1.23 MHz). Synchronization All base channels need to be synchronized to use CDMA. To provide synchronization, bases use the services of GPS (Global Positioning System), a satellite system that we discuss in the next section. Forward Transmission IS-95 has two different transmission techniques: one for use in the forward (base to mobile) direction and another for use in the reverse (mobile to base) direction. In the forward direction, communications between the base and all mobiles are synchronized; the base sends synchronized data to all mobiles. Each voice channel is digitized, producing data at a basic rate of 9.6 kbps. After adding error-correcting and repeating bits, and interleaving, the result is a signal of 19.2 ksps (kilosignals per second). This output is now scrambled using a 19.2-ksps signal. The scrambling signal is produced from a long code generator that uses the electronic serial number (ESN) of the mobile station and generates 242 pseudorandom chips, each chip having 42 bits. Note that the chips are generated pseudorandomly, not randomly, because the pattern repeats itself. The output of the long code generator is fed to a decimator, which chooses 1 bit out of 64 bits. The output of the decimator is used for scrambling. The scrambling is used to create privacy; the ESN is unique for each station.

The result of the scrambler is combined using CDMA. For each traffic channel, one Walsh 64 x 64 row chip is selected. The result is a signal of 1.228 Mcps (megachips per second).

19.2 ksps x 64 cps = 1.228 Mcps

The signal is fed into a QPSK modulator to produce a signal of 1.228 MHz. The resulting bandwidth is shifted appropriately, using FDMA. An analog channel creates64 digital channels, of which 55 channels are traffic channels (carrying digitized voice). Nine channels are used for control and synchronization:

##Channel 0 is a pilot channel. This channel sends a continuous stream of 1 s to mobile stations. The stream provides bit synchronization, serves as a phase reference for demodulation, and allows the mobile station to compare the signal strength of neighboring bases for handoff decisions.

##Channel 32 gives information about the system to the mobile station.

##Channels 1 to 7 are used for paging, to send messages to one or more mobile stations.

## Channels 8 to 31 and 33 to 63 are traffic channels carrying digitized voice from the base station to the corresponding mobile station.

Reverse Transmission The use of CDMA in the forward direction is possible because the pilot channel sends a continuous sequence of ls to synchronize transmission. The synchronization is not used in the reverse direction because we need an entity to do that, which is not feasible. Instead of CDMA, the reverse channels use DSSS (direct sequence spread spectrum), which we discussed in Chapter 8. Figure 16.11 shows a simplified diagram for reverse transmission.

Each voice channel is digitized, producing data at a rate of 9.6 kbps. However, after adding error-correcting and repeating bits, plus interleaving, the result is a signal of 28.8 ksps. The output is now passed through a 6/64 symbol modulaton The symbols are divided into six-symbol chunks, and each chunk is interpreted as a binary number (from 0 to 63). The binary number is used as the index to a 64 x 64 Walsh matrix for selection of a row of chips. Note that this procedure is not CDMA; each bit is not multiplied by the chips in a row. Each six-symbol chunk is replaced by a 64-chip code. This is done to provide a kind of orthogonality; it differentiates the streams of chips from the different mobile stations. The result creates a signal of 307.2 kbps or(28.8/6) x 64.Spreading is the next step; each chip is spread into 4. Again the ESN of the mobilestation creates a long code of 42 bits at a rate of 1.228 Mbps, which is 4 times 307.2. After spreading, each signal is modulated using QPSK, which is slightly different from the one used in the forward direction; we do not go into details here. Note that there is no multiple-access mechanism here; all reverse channels send their analog signal into the air, but the correct chips will be received by the base station due to spreading.

Although we can create 242 - 1 digital channels in the reverse direction (because of the long code generator), normally 94 channels are used; 62 are traffic channels, and 32 are channels used to gain access to the base station.

Two Data Rate Sets IS-95 defines two data rate sets, with four different rates in each set. The first set defines 9600, 4800, 2400, and 1200 bps. If, for example, the selected rate is 1200 bps, each bit is repeated 8 times to provide a rate of 9600 bps. The second set defines 14,400, 7200, 3600, and 1800 bps. This is possible by reducing the number of bits used for error correction. The bit rates in a set are related to the activity of the channel. If the channel is silent, only 1200 bits can be transferred, which improves the spreading by repeating each bit 8 times. Frequency-Reuse Factor In an IS-95 system, the frequency-reuse factor is normally 1 because the interference from neighboring cells cannot affect CDMA or DSSS transmission.

Wireless WAN

noreply@blogger.com (Utsav Basu) — Mon, 27 Apr 2009 22:57:22 PDT

Wireless WANs: Cellular
and Satellite Networks
Telephone

Wireless technology is also used in cellular telephony and satellite networks. We discuss the former in this chapter as well as examples of channelization access methods . We also briefly discuss satellite networks, a technology that eventually will be linked to cellular telephony to access the Internet directly.

CELLULAR TELEPHONY

Cellular telephony is designed to provide communications between two moving units, called mobile stations (MSs), or between one mobile unit and one stationary unit, often called a land unit. A service provider must be able to locate and track a caller, assign a channel to the call, and transfer the channel from base station to base station as the caller moves out of range. To make this tracking possible, each cellular service area is divided into small regions called cells. Each cell contains an antenna and is controlled by a solar or AC powered network station, called the base station (BS). Each base station, in turn, is controlled by a switching office, called a mobile switching center (MSC). The MSC coordinates communication between all the base stations and the telephone central office. It is a computerized center that is responsible for connecting calls, recording call information, and billing.Cell size is not fixed and can be increased or decreased depending on the population of the area. The typical radius of a cell is 1 to 12 mi. High-density areas require more, geographically smaller cells to meet traffic demands than do low-density areas. Once determined, cell size is optimized to prevent the interference of adjacent cell signals. The transmission power of each cell is kept low to prevent its signal from interfering with those of other cells.

Frequency-Reuse Principle

In general, neighboring cells cannot use the same set of frequencies for communication because it may create interference for the users located near the cell boundaries. However, the set of frequencies available is limited, and frequencies need to be reused. A frequency reuse pattern is a configuration of N cells, N being the reuse factor, in which each cell uses a unique set of frequencies. When the pattern is repeated, the frequencies can be reused. There are several different patterns.

Transmitting

To place a call from a mobile station, the caller enters a code of 7 or 10 digits (a phone number) and presses the send button. The mobile station then scans the band, seeking a setup channel with a strong signal, and sends the data (phone number) to the closest base station using that channel. The base station relays the data to the MSC. The MSC sends the data on to the telephone central office. If the called party is available, a connection is made and the result is relayed back to the MSC. At this point, the MSC assigns an unused voice channel to the call, and a connection is established. The mobile station automatically adjusts its tuning to the new channel, and communication can begin.

Receiving

When a mobile phone is called, the telephone central office sends the number to the MSC. The MSC searches for the location of the mobile station by sending query signals to each cell in a process called paging. Once the mobile station is found, the MSC transmits a ringing signal and, when the mobile station answers, assigns a voice channel to the call, allowing voice communication to begin.

Handoff

It may happen that, during a conversation, the mobile station moves from one cell to another. When it does, the signal may become weak. To solve this problem, the MSC monitors the level of the signal every few seconds. If the strength of the signal diminishes, the MSC seeks a new cell that can better accommodate the communication. The MSC then changes the channel carrying the call (hands the signal off from the old channel to a new one).

Hard Handoff Early systems used a hard handoff. In a hard handoff, a mobile station only communicates with one base station. When the MS moves from one cell to another, communication must first be broken with the previous base station before communication can be established with the new one. This nay create a rough transition.

Soft Handoff New systems use a soft handoff. In this case, a mobile station can communicate with two base stations at the same time. This means that, during handoff, a mobile station may continue with the new base station before breaking off from the old one.

Roaming

One feature of cellular telephony is called roaming. Roaming means, in principle, that a user can have access to communication or can be reached where there is coverage. A service provider usually has limited coverage. Neighboring service providers can provide extended coverage through a roaming contract. The situation is similar to snail mail between countries. The charge for delivery of a letter between two countries can be divided upon agreement by the two countries.

First Generation

Cellular teleph.ony is now in its second generation with the third on the horizon. The first generation was designed for voice communication using analog signals. We discuss one first-generation mobile system used in North America, AMPS.

AMPS

Advanced Mobile Phone System (AMPS) is one of the leading analog cellular systems in North America. It uses FDMA to separate channels in a link. Bands AMPS operates in the ISM 800-MHz band. The system uses two separate analog channels, one for forward (base station to mobile station) communication and one for reverse (mobile station to base station) communication. The band between 824 and 849 MHz carries reverse communication; the band between 869 and 894 MHz carries forward communication Each band is divided into 832 channels. However, two providers can share an area, which means 416 channels in each cell for each provider. Out of these 416, 21 channels are used for control, which leaves 395 channels. AMPS has a frequency reuse factor of 7; this means only one-seventh of these 395 traffic channels are actually available in a cell. Transmission AMPS uses FM and FSK for modulation. Figure 16.4 shows the trans- mission in the reverse direction. Voice channels are modulated using FM, and control channels use FSK to create 30-kHz analog signals. AMPS uses FDMA to divide each 25-MHz band into 30-kHz channels.

Second Generation

To provide higher-quality (less noise-prone) mobile voice communications, the second generation of the cellular phone network was developed. While the first generation was designed for analog voice communication, the second generation was mainly designed for digitized voice. Three major systems evolved in the second generation,

D-AMPS

The product of the evolution of the analog AMPS into a digital system is digital AMPS (D-AMPS). D-AMPS was designed to be backward-compatible with AMPS. This means that in a cell, one telephone can use AMPS and another D-AMPS. D-AMPS was first defined by IS-54 (Interim Standard 54) and later revised by IS-136. Band D-AMPS uses the same bands and channels as AMPS. Transmission Each voice channel is digitized using a very complex PCM and compression technique. A voice channel is digitized to 7.95 kbps. Three 7.95-kbps digital voice channels are combined using TDMA. The result is 48.6 kbps of digital data; much of this is overhead. As Figure 16.6 shows, the system sends 25 frames per second, with 1944 bits per frame. Each frame lasts 40 ms (1/25) and is divided into six slots shared by three digital channels; each channel is allotted two slots. Each slot holds 324 bits. However, only 159 bits comes from the digitized voice; 64 bits are for control and 101 bits are for error correction. In other words, each channel drops 159 bits of data into each of the two channels assigned to it. The system adds 64 control bits and 101 error-correcting bits. The resulting 48.6 kbps of digital data modulates a carrier using QPSK; the result is a 30-kHz analog signal. Finally, the 30-kHz analog signals share a 25-MHz band (FDMA). D-AMPS has a frequency reuse factor of 7.

BLUETOOTH

noreply@blogger.com (Utsav Basu) — Mon, 27 Apr 2009 22:51:38 PDT

BLUETOOTH

Bluetooth is a wireless LAN technology designed to connect devices of different functions such as telephones, notebooks, computers (desktop and laptop), cameras, printers, coffee makers, and so on. A Bluetooth LAN is an ad hoc network, which means that the network is formed spontaneously; the devices, sometimes called gadgets, find each other and make a network called a piconet. A Bluetooth LAN can even be connected to the Internet if one of the gadgets has this capability. A Bluetooth LAN, by nature, can not be large. If there are many gadgets that try to connect, there is chaos. Bluetooth technology has several applications. Peripheral devices such as a wireless mouse or keyboard can communicate with the computer through this technology. Monitoring devices can communicate with sensor devices in a small health care center. Home security devices can use this technology to connect different sensors to the main security controller. Conference attendees can synchronize their laptop computers at a conference. Bluetooth was originally started as a project by the Ericsson Company. It is named for Harald Blaatand the king of Denmark (940-981) who united Denmark and Norway.

Blaatand translates to Bluetooth in English.

Today, Bluetooth technology is the implementation of a protocol defined by the
IEEE 802.15 standard. The standard defines a wireless personal-area network (PAN)
operable in an area the size of a room or a hall.

Architecture

Bluetooth defines two types of networks: piconet and scatternet.

Piconets

A Bluetooth network is called a piconet, or a small net. A piconet can have up to eight stations, one of which is called the primary; ? the rest are called secondaries. All the secondary stations synchronize their clocks and hopping sequence with the primary.Note that a piconet can have only one primary station. The communication between the primary and the secondary can be one-to-one or one-to-many.

Although a piconet can have a maximum of seven secondaries, an additional eight secondaries can be in the parked state. A secondary in a parked state is synchronize with the primary, but cannot take part in communication until it is moved from the parked state. Because only eight stations can be active in a piconet, activating a station from the parked state means that an active station must go to the parked state.

Scatternet

Piconets can be combined to form what is called a scatternet. A secondary station intone piconet can be the primary in another piconet. This station can receive messages from the primary in the first piconet (as a secondary) and, acting as a primary, deliver them to secondaries in the second piconet. A station can be a member of two piconets.

Bluetooth Devices

A Bluetooth device has a built-in short-range radio transmitter. The current data rate is 1 Mbps with a 2.4-GHz bandwidth. This means that there is a possibility of interference between the IEEE 802.1 lb wireless LANs and Bluetooth LANs.

Radio Layer

The radio layer is roughly equivalent to the physical layer of the Internet model. Bluetooth devices are low-power and have a range of 10 m.

Band

Bluetooth uses a 2.4-GHz ISM band divided into 79 channels of 1 MHz each.

FHSS

Bluetooth uses the frequency-hopping spread spectrum (FHSS) method in the physical layer to avoid interference from other devices or other networks. Bluetooth hops 1600 times per second, which means that each device changes its modulation frequency 1600 times per second. A device uses a frequency for only 625 gs (1/1600 s) before it hops to another frequency; the dwell time is 625 gs.

Network Allocation Vector

noreply@blogger.com (Utsav Basu) — Mon, 27 Apr 2009 22:46:40 PDT

Network Allocation Vector How do other stations defer sending their data if one station acquires access? In other words, how is the collision avoidance aspect of this protocol accomplished ? The key is a feature called NAV. When a station sends an RTS frame, it includes the duration of time that it needs to occupy the channel. The stations that are affected by this transmission create a timer called a network allocation vector (NAV) that shows how much time must pass before these stations are allowed to check the channel for idleness. Each time a station accesses the system and sends an RTS frame, other stations start their NAV. In other words, each station, before sensing the physical medium to see if it is idle, first checks its NAV to see if it has expired.

Collision During Handshaking What happens if there is collision during the time when RTS or CTS control frames are in transition, often called the handshaking period? Two or more stations may try to send RTS frames at the same time. These control frames may collide. However, because there is no mechanism for collision detection, the sender assumes there has been a collision if it has not received a CTS frame from the receiver. The back-off strategy is employed, and the sender tries again.

Point Coordination Function (PCF)

The point coordination function (PCF) is an optional access method that can be implemented in an infrastructure network (not in an ad hoc network). It is implemented on top of the DCF and is used mostly for time-sensitive transmission. PCF has a centralized, contention-free polling access method. The AP performs polling for stations that are capable of being polled. The stations are polled one after another, sending any data they have to the AP. To give priority to PCF over DCF, another set of interframe spaces has been defined: PIFS and SIFS. The SIFS is the same as that in DCF, but the PIFS (PCF IFS) is shorter than the DIFS. This means that if, at the same time, a station wants to use only DCF and an AP wants to use PCF, the AP has priority. Due to the priority of PCF over DCF, stations that only use DCF may not gain access to the medium. To prevent this, a repetition interval has been designed to cover both contention-free (PCF) and contention-based (DCF) traffic. The repetition interval, which is repeated continuously, starts with a special control frame, called a beacon frame. When the stations hear the beacon frame, they start their NAV for the duration of the contention-free period of the repetition interval. During the repetition interval, the PC (point controller) can send a poll frame,
receive data, send an ACK, receive an ACK, or do any combination of these (802.11 uses piggybacking). At the end of the contention-free period, the PC sends a CF end (contention-free end) frame to allow the contention-based stations to use the medium.

Fragnentation

The wireless environment is very noisy; a corrupt frame has to be retransmitted. The protocol, therefore, recommends fragmentation--the division of a large frame into smaller ones. It is more efficient to resend a small frame than a large one.

Wireless, Wireless LANs, LAN Arcitecture,IEEE 802.11

noreply@blogger.com (Utsav Basu) — Sun, 02 Aug 2009 22:00:40 PDT

What is Wireless LAN, Wireless LAN overview.

The wireless LAN (WLAN) is a wireless local area network that communicate two or more computers or devices using spread-spectrum or OFDM modulation technology based to enable Links between devices in a limited area or local area. That helps users to move in mobility around within a broad coverage area and still be connected to the network.

Easy Installation system make the WLAN very popular for the Home users, and For its mobility features It is best for the Laptop users. Public businesses Like shops, coffe shops, malls have begun to offer wireless access to their customers; some are even provided as a free service. Large wireless network projects are being put up in many major cities: New York City, Salt lake city in INDIA for instance, has begun a pilot program to cover all five boroughs of the city with wireless Internet access.

Source : wikipedia

Wireless LANs

Wireless communication is one of the fastest-growing technologies. The demand for connecting devices without the use of cables is increasing everywhere. Wireless LANs can be found on college campuses, in office buildings, and in many public areas. In this chapter, we concentrate on two promising wireless technologies for LANs: IEEE 802.11 wireless LANs, sometimes called wireless Ethernet, and Bluetooth, a technology for small wireless LANs. Although both protocols need several layers to operate, we concentrate mostly on the physical and data link layers.

IEEE 802.11

IEEE has defined the specifications for a wireless LAN, called IEEE 802.11, which covers the physical and data link layers.

Architecture

The standard defines two kinds of services: the basic service set (BSS) and the extended service set (ESS). Basic Service Set IEEE 802.1 ] defines the basic service set (BSS) as the building block of a wireless LAN. A basic service set is made of stationary or mobile wireless stations and an optional central base station, known as the access point (AP).

The BSS without an AP is a stand-alone network and cannot send data to other BSSs. It is called an ad hoc architecture. In this architecture, stations can form a network without the need of an AP; they can locate one another and agree to be part of a BSS. A BSS with an AP is sometimes referred to as an infrastructure network.

Extended Service Set

An extended service set (ESS) is made up of two or more BSSs with APs. In this case, the BSSs are connected through a distribution system, which is usually a wired LAN. The distribution system connects the APs in the BSSs. IEEE 802.11 does not restrict the distribution system; it can be any IEEE LAN such as an Ethernet. Note that the extended service set uses two types of stations: mobile and stationary. The mobile stations are normal stations inside a BSS. The stationary stations are AP stations that are part of a wired LAN. When BSSs are connected, the stations within reach of one another can communicate without the use of an AP. However, communication between two stations in two different BSSs usually occurs via two APs. The idea is similar to communication in a cellular network if we consider each BSS to be a cell and each AP to be a base station. Note that a mobile station can belong to more than one BSS at the same time.

Station Types
IEEE 802.11 defines three types of stations based on their mobility in a wireless LAN: no-transition, BSS-transition, and ESS-transition mobility. A station with no-transition mobility is either stationary (not moving) or moving only inside a BSS. A station with BSS-transition mobility can move from one BSS to another, but the movement is confined inside one ESS. A station with ESS-transition mobility can move from one ESS to another. However, IEEE 802.11 does not guarantee that communication is continuous during the move.

MAC Sublayer

IEEE 802.11 defines two MAC sublayers: the distributed coordination function (DCF) and point coordination function (PCF).

Benifits of Wireless LANs -

Today The use and the popularity of Wireless Lan is Increasing very fast, Why Everyone is using such a System Let us see -

Convenience : For its Wireless facility Every one can access this network from anywhere or any convenient location or Home and office. For the Laptop Style computers This is best and relevant.

Mobility :Mobility features is the best features of WLANs, Any one can use this network in any where like Coffee shops, Shopping Mall, User can use it outside of their normal work place. This WLANs are cost effective also.

CDMA

noreply@blogger.com (Utsav Basu) — Mon, 27 Jul 2009 20:56:01 PDT

Code-Division Multiple Access (CDMA)

Code-division multiple access (CDMA) was conceived several decades ago. Recent advances in electronic technology have finally made its implementation possible. CDMA differs from FDMA because only one channel occupies the entire bandwidth of the link. It differs from TDMA because all stations can send data simultaneously; there is no timesharing.

Analogy

Let us first give an analogy. CDMA simply means communication with different codes. For example, in a large room with many people, two people can talk in English if nobody else understands English. Another two people can talk in Chinese if they are the only ones who understand Chinese, and so on. In other words, the common channel, the space of the room in this case, can easily allow communication between several couples, but in different languages (codes).

Idea

Let us assume we have four stations 1, 2, 3, and 4 connected to the same channel. The data from station 1 are d 1, from station 2 are d 2, and so on. The code assigned to the first station is cl, to the second is c2, and so on. We assume that the assigned codes have two properties.

1. If we multiply each code by another, we get 0.
2. If we multiply each code by itself, we get 4 (the number of stations).

With these two properties in mind, let us see how the above four stations can send data using the same common channel,

data that go on the channel are the sum of all these terms, as shown in the box. Any station that wants to receive data from one of the other three multiplies the data on the channel by the code of the sender. For example, suppose stations 1 and 2 are talking to each other. Station 2 wants to hear what station 1 is saying. It multiplies the data on the channel by c 1, the code of station 1.

Because (c 1 ?? Cl) is 4, but (c 2 ?? Cl), (c. Cl), and (c 4 - c 1) are all Os, station 2 divides
the result by 4 to get the data from station 1.
data = (d 1 - c t + d 2 ?? c 2 +d 3 - c 3 + d 4- c4) ?? c I
=d l.c 1.c l+d 2.c 2.c l+d 3-c 3.c l+d 4-c4.c l=4Xd

Chips

CDMA is based on coding theory. Each station is assigned a code, which is a sequence of numbers called chips.

CHANNELIZATION

noreply@blogger.com (Utsav Basu) — Sat, 25 Apr 2009 09:13:20 PDT

CHANNELIZATION

Channelization is a multiple-access method in which the available bandwidth of a link is shared in time, frequency, or through code, between different stations. In this section, we discuss three channelization protocols: FDMA, TDMA, and CDMA.

Frequency-Division Multiple Access (FDMA)

In frequency-division multiple access (FDMA), the available bandwidth is divided into frequency bands. Each station is allocated a band to send its data. In other words, each band is reserved for a specific station, and it belongs to the station all the time. Each station also uses a bandpass filter to confine the transmitter frequencies. To prevent station interferences, the allocated bands are separated from one another by smallguard bands.

FDMA specifies a predetermined frequency band for the entire period of communication. This means that stream data (a continuous flow of data that may not be packetized) can easily be used with FDMA. We will see in Chapter 16 how this feature can be used in cellular telephone systems.

We need to emphasize that although FDMA and FDM conceptually seem similar,there are differences between them. FDM, is a physical layertechnique that combines the loads from low-bandwidth channels and transmits them by using a high-bandwidth channel. The channels that are combined are low-pass. The multiplexer modulates the signals, combines them, and creates a bandpass signal. The bandwidth of each channel is shifted by the multiplexer. FDMA, on the other hand, is an access method in the data link layer. The data link layer in each station tells its physical layer to make a bandpass signal from the data passed to it. The signal must be created in the allocated band. There is no physical multiplexer at the physical layer. The signals created at each station are automatically bandpass-filtered. They are mixed when they are sent to the common channel.

Time-Division Multiple Access (TDMA)

In time-division multiple access (TDMA), the stations share the bandwidth of the channel in time. Each station is allocated a time slot during which it can send data. Each station transmits its data in is assigned time slot. The main problem with TDMA lies in achieving synchronization between the different stations. Each station needs to know the beginning of its slot and the location of its slot. This may be difficult because of propagation delays introduced in the system if the stations are spread over a large area. To compensate for the delays, we can insert guard times. Synchronization is normally accomplished by having some synchronization bits (normally referred to as preamble bits) at the beginning of each slot. We also need to emphasize that although TDMA and TDM conceptually seem the same, there are differences between them. TDM, is a physical layer technique that combines the data from slower channels and transmits them by using a faster channel. The process uses a physical multiplexer that interleaves data units from each channel. TDMA, on the other hand, is an access method in the data link layer. The data link layer in each station tells its physical layer to use the allocated time slot. There is no physical multiplexer at the physical layer.

Token Passing

noreply@blogger.com (Utsav Basu) — Sat, 25 Apr 2009 09:09:55 PDT

Token Passing

In the token-passing method, the stations in a network are organized in a logical ring. In other words, for each station, there is a predecessor and a successor. The predecessor is the station which is logically before the station in the ring; the successor is the station which is after the station in the ring. The current station is the one that is accessing the channel now. The fight to this access has been passed from the predecessor to the current station. The right will be passed to the successor when the current station has no more data to send. But how is the right to access the channel passed from one station to another? In this method, a special packet called a token circulates through the ring. The possession of the token gives the station the right to access the channel and send its data. When a station has some data to send, it waits until it receives the token from its predecessor. It then holds the token and sends its data. When the station has no more data to send, it releases the token, passing it to the next logical station in the ring. The station cannot send data until it receives the token again in the next round. In this process, when a station receives the token and has no data to send, it just passes the data to the
next station. Token management is needed for this access method. Stations must be limited in the time they can have possession of the token. The token must be monitored to ensure it has not been lost or destroyed. For example, if a station that is holding the token fails, the token will disappear from the network. Another function of token management is to assign priorities to the stations and to the types of data being transmitted. And finally, token management is needed to make low-priority stations release the token to highpriority stations.

Logical Ring

In a token-passing network, stations do not have to be physically connected in a ring; the ring can be a logical one. In the physical ring topology, when a station sends the token to its successor, the token cannot be seen by other stations; the successor is the next one in line. This means that the token does not have to have the address of the next successor. The problem with this topology is that if one of the links--the medium between two adjacent stations-- fails, the whole system fails. The dual ring topology uses a second (auxiliary) ring which operates in the reverse
direction compared with the main ring. The second ring is for emergencies only (such as a spare tire for a car). If one of the links in the main ring fails, the system automatically combines the two rings to form a temporary ring. After the failed link is restored, the auxiliary ring becomes idle again. Note that for this topology to work, each station needs to have two transmitter ports and two receiver ports. The high-speed Token Ring networks called FDDI (Fiber Distributed Data Interface) and CDDI (Copper Distributed Data Interface) use this topology. In the bus ring topology, also called a token bus, the stations are connected to a single cable called a bus. They, however, make a logical ring, because each station knows the address of its successor (and also predecessor for token management purposes). When a station has finished sending its data, it releases the token and inserts the address of its successor in the token. Only the station with the address matching the destination address of the token gets the token to access the shared media. The Token Bus LAN, standardized by IEEE, uses this topology. In a star ring topology, the physical topology is a star. There is a hub, however, that acts as the connector. The wiring inside the hub makes the ring; the stations are connected to this ring through the two wire connections. This topology makes the network less prone to failure because if a link goes down, it will be bypassed by the hub and the rest of the stations can operate. Also adding and removing stations from the ring is easier. This topology is still used in the Token Ring LAN designed

CSMA/CA

noreply@blogger.com (Utsav Basu) — Sat, 25 Apr 2009 09:07:11 PDT

CSMA/CA and Wireless Networks

CSMA/CA was mostly intended for use in wireless networks. The procedure described above, however, is not sophisticated enough to handle some particular issues related to wireless networks, such as hidden terminals or exposed terminals. We will see how these issues are solved by augmenting the above protocol with hand-shaking features.

CONTROLLED ACCESS

In controlled access, the stations consult one another to find which station has the right to send. A station cannot send unless it has been authorized by other stations. We discuss three popular controlled-access methods.

Reservation

In the reservation method, a station needs to make a reservation before sending data. Time is divided into intervals. In each interval, a reservation frame precedes the data frames sent in that interval.

Polling

Polling works with topologies in which one device is designated as a primary station and the other devices are secondary stations. All data exchanges must be made through the primary device even when the ultimate destination is a secondary device. The primary device controls the link; the secondary devices follow its instructions. It is up to the primary device to determine which device is allowed to use the channel at a given time. The primary device, therefore, is always the initiator of a session.

If the primary wants to receive data, it asks the secondaries if they have anything to send; this is called poll function. If the primary wants to send data, it tells the secondary to get ready to receive; this is called select function.

Select

The select function is used whenever the primary device has something to send. Remember that the primary controls the link. If the primary is neither sending nor receiving data, it knows the link is available. If it has something to send, the primary device sends it. What it does not know,
however, is whether the target device is prepared to receive. So the primary must alert the secondary to the upcoming transmission and wait for an acknowledgment of the secondary's ready status. Before sending data, the primary creates and transmits a select (SEL) frame, one field of which includes the address of the intended secondary.

Poll

The poll function is used by the primary device to solicit transmissions from the secondary devices. When the primary is ready to receive data, it must ask (poll) each device in turn if it has anything to send. When the first secondary is approached, it responds either with a NAK frame if it has nothing to send or with data (in the form of a data frame) if it does. If the response is negative (a NAK frame), then the primary polls the next secondary in the same manner until it finds one with data to send. When the response is positive (a data frame), the primary reads the frame and returns an acknowledgment (ACK frame), verifying its receipt.

Collision Avoidance

noreply@blogger.com (Utsav Basu) — Sat, 25 Apr 2009 09:04:25 PDT

Carrier Sense Multiple Access with
Collision Avoidance (CSMA/CA)

The basic idea behind CSMA/CD is that a station needs to be able to receive while transmitting to detect a collision. When there is no collision, the station receives one signal: its own signal. When there is a collision, the station receives two signals: its own signal and the signal transmitted by a second station. To distinguish between these two cases, the received signals in these two cases must be significantly different. In other words, the signal from the second station needs to add a significant amount of energy to the one created by the first station. In a wired network, the received signal has almost the same energy as the sent signal because either the length of the cable is short or there are repeaters that amplify the energy between the sender and the receiver. This means that in a collision, the detected energy almost doubles. However, in a wireless network, much of the sent energy is lost in transmission. The received signal has very little energy. Therefore, a collision may add only 5 to 10 percent additional energy. This is not useful for effective collision detection. We need to avoid collisions on wireless networks because they cannot be detected. Carder sense multiple access with collision avoidance (CSMA/CA) was invented for this network. Collisions are avoided through the use of CSMA/CA's three strategies: the interframe space, the contention window, and acknowledgments.

Interframe Space (IFS)

First, collisions are avoided by deferring transmission even if the channel is found idle. When an idle channel is found, the station does not send immediately. It waits for a period of time called the interframe space or IFS. Even though the channel may appear idle when it is sensed, a distant station may have already started transmitting. The distant station's signal has not yet reached this station. The IFS time allows the front of the transmitted signal by the distant station to reach this station. If after the IFS time the channel is still idle, the station can send, but it still needs to wait a time equal to the contention time (described next). The IFS variable can also be used to prioritize stations or frame types. For example, a station that is assigned a shorter IFS has a higher priority.

Contention Window

The contention window is an amount of time divided into slots. A station that is ready to send chooses a random number of slots as its wait time. The number of slots in the window changes according to the binary exponential back-off strategy. This means that it is set to one slot the first time and then doubles each time the station cannot detect an idle channel after the IFS time. This is very similar to the p-persistent method except that a random outcome defines the number of slots taken by the waiting station. One interesting point about the contention window is that the station needs to sense the channel after each time slot. However, if the station finds the channel busy, it does not restart the process; it just stops the timer and restarts it when the channel is sensed as idle. This gives priority to the station with the longest waiting time.

Acknowledgment

With all these precautions, there still may be a collision resulting in destroyed data. In addition, the data may be corrupted during the transmission. The positive acknowledgment and the time-out timer can help guarantee that the receiver has received the frame.

Procedure

Note that the channel needs to be sensed before and after the IFS. The channel also needs to be sensed during the contention time. For each time slot of the contention window, the channel is sensed. If it is found idle, the timer continues; if the channel is found busy, the timer is stopped and continues after the timer becomes idle again.

Network fundamental

IP,IPv6 Routing Protocols, Internet protocols version six, IPv6

Table 4-1. Taxonomy of Routing Protocols

IP Mobility, IP, IPv6, IP version 6, Internet protocol

IP Mobility

Data Cabling Rules,Reliable Cabling, Poor Cabling Cost

Network Fundamental and its Components

How to Setup a LAN

Dial UP Technology

Telecommunications Systems

Telecommunications Concepts

Frame Relay Layers

Virtual-Circuit Networks

SONET NETWORKS

Encapsulation

STS-l frame: line overhead

SONET LAYERS

SONET/SDH

GSM

Wireless WAN

BLUETOOTH

Network Allocation Vector

Wireless, Wireless LANs, LAN Arcitecture,IEEE 802.11

CDMA

CHANNELIZATION

Token Passing

CSMA/CA

Collision Avoidance