What is a computer Network?
A network is any collection of independent computers that communicate with one another over a shared network medium.A computer network is a collection of two or more connected computers. When these computers are joined in a network, people can share files and peripherals such as modems, printers, tape backup drives, or CD-ROM drives. When networks at multiple locations are connected using services available from phone companies, people can send e-mail, share links to the global Internet, or conduct video conferences in real time with other remote users. As companies rely on applications like electronic mail and database management for core business operations, computer networking becomes increasingly more important.
Every network includes:
- · At least two computers Server or Client workstation.
- · Networking Interface Card's (NIC)
- · A connection medium, usually a wire or cable, although wireless communication between networked computers and peripherals is also possible.
- · Network Operating system software, such as Microsoft Windows NT or 2000, Novell NetWare, Unix and Linux.
Types of Networks:
LANs (Local Area Networks)
A network is any collection of independent computers that communicate with one another over a shared network medium. LANs are networks usually confined to a geographic area, such as a single building or a college campus. LANs can be small, linking as few as three computers, but often link hundreds of computers used by thousands of people. The development of standard networking protocols and media has resulted in worldwide proliferation of LANs throughout business and educational organizations.
WANs (Wide Area Networks)
Wide area networking combines multiple LANs that are geographically separate. This is accomplished by connecting the different LANs using services such as dedicated leased phone lines, dial-up phone lines (both synchronous and asynchronous), satellite links, and data packet carrier services. Wide area networking can be as simple as a modem and remote access server for employees to dial into, or it can be as complex as hundreds of branch offices globally linked using special routing protocols and filters to minimize the expense of sending data sent over vast distances.
Internet
The Internet is a system of linked networks that are worldwide in scope and facilitate data communication services such as remote login, file transfer, electronic mail, the World Wide Web and newsgroups.
With the meteoric rise in demand for connectivity, the Internet has become a communications highway for millions of users. The Internet was initially restricted to military and academic institutions, but now it is a full-fledged conduit for any and all forms of information and commerce. Internet websites now provide personal, educational, political and economic resources to every corner of the planet.
With the meteoric rise in demand for connectivity, the Internet has become a communications highway for millions of users. The Internet was initially restricted to military and academic institutions, but now it is a full-fledged conduit for any and all forms of information and commerce. Internet websites now provide personal, educational, political and economic resources to every corner of the planet.
Intranet
With the advancements made in browser-based software for the Internet, many private organizations are implementing intranets. An intranet is a private network utilizing Internet-type tools, but available only within that organization. For large organizations, an intranet provides an easy access mode to corporate information for employees.
MANs (Metropolitan area Networks)
The refers to a network of computers with in a City.
VPN (Virtual Private Network)
VPN uses a technique known as tunneling to transfer data securely on the Internet to a remote access server on your workplace network. Using a VPN helps you save money by using the public Internet instead of making long–distance phone calls to connect securely with your private network. There are two ways to create a VPN connection, by dialing an Internet service provider (ISP), or connecting directly to Internet.
Categories of Network:
Network can be divided in to two main categories:
- Peer-to-peer.
- Server – based.
In peer-to-peer networking there are no dedicated servers or hierarchy among the computers. All of the computers are equal and therefore known as peers. Normally each computer serves as Client/Server and there is no one assigned to be an administrator responsible for the entire network.
Peer-to-peer networks are good choices for needs of small organizations where the users are allocated in the same general area, security is not an issue and the organization and the network will have limited growth within the foreseeable future.
The term Client/server refers to the concept of sharing the work involved in processing data between the client computer and the most powerful server computer.
Peer-to-peer networks are good choices for needs of small organizations where the users are allocated in the same general area, security is not an issue and the organization and the network will have limited growth within the foreseeable future.
The term Client/server refers to the concept of sharing the work involved in processing data between the client computer and the most powerful server computer.
The client/server network is the most efficient way to provide:
- Databases and management of applications such as Spreadsheets, Accounting, Communications and Document management.
- Network management.
- Centralized file storage.
The client/server model is basically an implementation of distributed or cooperative processing. At the heart of the model is the concept of splitting application functions between a client and a server processor. The division of labor between the different processors enables the application designer to place an application function on the processor that is most appropriate for that function. This lets the software designer optimize the use of processors--providing the greatest possible return on investment for the hardware.
Client/server application design also lets the application provider mask the actual location of application function. The user often does not know where a specific operation is executing. The entire function may execute in either the PC or server, or the function may be split between them. This masking of application function locations enables system implementers to upgrade portions of a system over time with a minimum disruption of application operations, while protecting the investment in existing hardware and software.
Client/server application design also lets the application provider mask the actual location of application function. The user often does not know where a specific operation is executing. The entire function may execute in either the PC or server, or the function may be split between them. This masking of application function locations enables system implementers to upgrade portions of a system over time with a minimum disruption of application operations, while protecting the investment in existing hardware and software.
The OSI Model:
Open System Interconnection (OSI) reference model has become an International standard and serves as a guide for networking. This model is the best known and most widely used guide to describe networking environments. Vendors design network products based on the specifications of the OSI model. It provides a description of how network hardware and software work together in a layered fashion to make communications possible. It also helps with trouble shooting by providing a frame of reference that describes how components are supposed to function.
DNS (Domain Name Service)
The Domain Name System (DNS) is a system that stores information associated with domain names in a distributed database on networks, such as the Internet. The domain name system associates many types of information with domain names, but most importantly, it provides the IP address associated with the domain name. It also lists mail exchange servers accepting e-mail for each domain.
DNS is useful for several reasons. Most well known, the DNS makes it possible to attach hard-to-remember IP addresses (such as 207.142.131.206) to easy-to-remember domain names (such as "wikipedia.org") Humans take advantage of this when they recite URLs and e-mail addresses. Less recognized, the domain name system makes it possible for people to assign authoritative names, without needing to communicate with a central registrar each time.
The domain name space is a gigantic tree of domain names. Each node or leaf in the tree is associated with resource records, which hold the information associated with the domain name. The tree is divided into zones. A zone is a collection of connected nodes that are authoritatively served by an authoritative DNS nameserver. (Note that a single nameserver can host several zones.)
When a system administrator wants to let another administrator control a part of the domain name space within his or her zone of authority, he or she can delegate control to the other administrator. This splits a part of the old zone off into a new zone, which is served by the second administrator's nameservers. The old zone is no longer authoritative for what is under the authority of the new zone.
The information associated with nodes is looked up by a resolver. A resolver knows how to communicate with name servers by sending DNS requests, and heeding DNS responses. Resolving usually entails recursing through several name servers to find the needed information. Some resolvers are simple, and can only communicate with a single ame server. These simple resolvers rely on a recursing name server to perform the work of finding information for it.
Important categories of data stored in the DNS include an A record or address record maps a hostname to its 32-bit IPv4 address, an AAAA record or IPv6 address record maps a hostname to its 128-bit IPv6 address, a CNAME record or canonical name record makes one domain name an alias of another. The aliased domain gets all the subdomains and DNS records of the original, an MX record or mail exchange record maps a domain name to a list of mail exchange servers for that domain, a PTR record or pointer record maps an IPv4 address to the canonical name for that host. Setting up a PTR record for a hostname in the in-addr.arpa domain that corresponds to an IP address implements reverse DNS lookup for that address. For example (at the time of writing), www.icann.net has the IP address 192.0.34.164, but a PTR record maps 164.34.0.192.in-addr.arpa to its canonical name, referrals.icann.org., an NS record or name server record maps a domain name to a list of DNS servers for that domain. Delegations depend on NS records, an SOA record or start of authority record specifies the DNS server providing authoritative information about an Internet domain, an SRV record is a generalized service location record, a TXT record allows an administrator to insert arbitrary text into a DNS record. For example, this record is used to implement the Sender Policy Framework specification.
DNS is useful for several reasons. Most well known, the DNS makes it possible to attach hard-to-remember IP addresses (such as 207.142.131.206) to easy-to-remember domain names (such as "wikipedia.org") Humans take advantage of this when they recite URLs and e-mail addresses. Less recognized, the domain name system makes it possible for people to assign authoritative names, without needing to communicate with a central registrar each time.
The domain name space is a gigantic tree of domain names. Each node or leaf in the tree is associated with resource records, which hold the information associated with the domain name. The tree is divided into zones. A zone is a collection of connected nodes that are authoritatively served by an authoritative DNS nameserver. (Note that a single nameserver can host several zones.)
When a system administrator wants to let another administrator control a part of the domain name space within his or her zone of authority, he or she can delegate control to the other administrator. This splits a part of the old zone off into a new zone, which is served by the second administrator's nameservers. The old zone is no longer authoritative for what is under the authority of the new zone.
The information associated with nodes is looked up by a resolver. A resolver knows how to communicate with name servers by sending DNS requests, and heeding DNS responses. Resolving usually entails recursing through several name servers to find the needed information. Some resolvers are simple, and can only communicate with a single ame server. These simple resolvers rely on a recursing name server to perform the work of finding information for it.
Important categories of data stored in the DNS include an A record or address record maps a hostname to its 32-bit IPv4 address, an AAAA record or IPv6 address record maps a hostname to its 128-bit IPv6 address, a CNAME record or canonical name record makes one domain name an alias of another. The aliased domain gets all the subdomains and DNS records of the original, an MX record or mail exchange record maps a domain name to a list of mail exchange servers for that domain, a PTR record or pointer record maps an IPv4 address to the canonical name for that host. Setting up a PTR record for a hostname in the in-addr.arpa domain that corresponds to an IP address implements reverse DNS lookup for that address. For example (at the time of writing), www.icann.net has the IP address 192.0.34.164, but a PTR record maps 164.34.0.192.in-addr.arpa to its canonical name, referrals.icann.org., an NS record or name server record maps a domain name to a list of DNS servers for that domain. Delegations depend on NS records, an SOA record or start of authority record specifies the DNS server providing authoritative information about an Internet domain, an SRV record is a generalized service location record, a TXT record allows an administrator to insert arbitrary text into a DNS record. For example, this record is used to implement the Sender Policy Framework specification.
SMTP (Simple Mail Transfer Protocol)
Simple Mail Transfer Protocol (SMTP) is the de facto standard for email transmission across the Internet. SMTP is a relatively simple, text-based protocol, where one or more recipients of a message are specified (and in most cases verified to exist) and then the message text is transferred. It is quite easy to test a SMTP server using the telnet program. SMTP uses TCP port 25. To determine the SMTP server for a given domain name, the MX (Mail eXchange) DNS record is used. SMTP started becoming widely used in the early 1980s. At the time, it was a complement to UUCP which was better suited to handle e-mail transfers between machines that were intermittently connected. SMTP, on the other hand, works best when both the sending and receiving machines are connected to the network all the time. Send mail was one of the first (if not the first) mail transfer agents to implement SMTP. As of 2001 there are at least 50 programs that implement SMTP as a client (sender of messages) or a server (receiver of messages). Some other popular SMTP server programs include Philip Hazel's exim, IBM's Postfix, D. J. Bernstein's qmail, and Microsoft Exchange Server.
This protocol started out as purely ASCII text-based, it did not deal well with binary files. Standards such as MIME were developed to encode binary files for transfer through SMTP. Today, most SMTP servers support the 8BITMIME extension, permitting binary files to be transmitted almost as easily as plain text.
SMTP is a "push" protocol that does not allow one to "pull" messages from a remote server on demand. To do this a mail client must use POP3 or IMAP. Another SMTP server can trigger a delivery in SMTP using ETRN.
One of the limitations of the original SMTP is that it has no facility for authentication of senders. Therefore the SMTP-AUTH extension was defined.
In spite of this, E-mail spamming is still a major problem. Modifying SMTP extensively, or replacing it completely, is not believed to be practical, due to the network effects of the huge installed base of SMTP. Internet Mail 2000 is one such proposal for replacement.For this reason, there are a number of proposals for sideband protocols that will assist SMTP operation. The Anti-Spam Research Group of the IRTF is working on a number of Email authentication and other proposals for providing simple source authentication that is flexible, lightweight, and scalable.
After establishing a connection between the sender (the client) and the receiver (the server), the following is a legal SMTP session. In the following conversation, everything sent by the client is prefaced with "C:" and everything sent by the server is prefaced with "S:". On most computer systems, a connection can be established using the telnet command on the sending machine, for example telnet www.example.com 25 which opens an SMTP connection from the sending machine to the host www.example.com.
S: 220 www.example.com ESMTP Postfix
C: HELO mydomain.com
S: 250 Hello mydomain.com
C: MAIL FROM:
S: 250 Ok
C: RCPT TO:
S: 250 Ok
C: DATA
S: 354 End data with .
C: Subject: test message
C: From: sender@mydomain.com
C: To: friend@example.com
C:
C: Hello.
C: This is a test.
C: Goodbye.
C: .
S: 250 Ok: queued as 12345
C: QUIT
S: 221 Bye
Although optional and not shown above, nearly all clients ask the server which SMTP extensions the server supports by using the EHLO greeting. These clients use HELO only if the server does not respond to EHLO.
This protocol started out as purely ASCII text-based, it did not deal well with binary files. Standards such as MIME were developed to encode binary files for transfer through SMTP. Today, most SMTP servers support the 8BITMIME extension, permitting binary files to be transmitted almost as easily as plain text.
SMTP is a "push" protocol that does not allow one to "pull" messages from a remote server on demand. To do this a mail client must use POP3 or IMAP. Another SMTP server can trigger a delivery in SMTP using ETRN.
One of the limitations of the original SMTP is that it has no facility for authentication of senders. Therefore the SMTP-AUTH extension was defined.
In spite of this, E-mail spamming is still a major problem. Modifying SMTP extensively, or replacing it completely, is not believed to be practical, due to the network effects of the huge installed base of SMTP. Internet Mail 2000 is one such proposal for replacement.For this reason, there are a number of proposals for sideband protocols that will assist SMTP operation. The Anti-Spam Research Group of the IRTF is working on a number of Email authentication and other proposals for providing simple source authentication that is flexible, lightweight, and scalable.
After establishing a connection between the sender (the client) and the receiver (the server), the following is a legal SMTP session. In the following conversation, everything sent by the client is prefaced with "C:" and everything sent by the server is prefaced with "S:". On most computer systems, a connection can be established using the telnet command on the sending machine, for example telnet www.example.com 25 which opens an SMTP connection from the sending machine to the host www.example.com.
S: 220 www.example.com ESMTP Postfix
C: HELO mydomain.com
S: 250 Hello mydomain.com
C: MAIL FROM:
S: 250 Ok
C: RCPT TO:
S: 250 Ok
C: DATA
S: 354 End data with .
C: Subject: test message
C: From: sender@mydomain.com
C: To: friend@example.com
C:
C: Hello.
C: This is a test.
C: Goodbye.
C: .
S: 250 Ok: queued as 12345
C: QUIT
S: 221 Bye
Although optional and not shown above, nearly all clients ask the server which SMTP extensions the server supports by using the EHLO greeting. These clients use HELO only if the server does not respond to EHLO.
HTTP (HyperText Transfer Protocol)
HyperText Transfer Protocol (HTTP) is the primary method used to convey information on the World Wide Web. The original purpose was to provide a way to publish and receive HTML pages. Development of HTTP was coordinated by the World Wide Web Consortium and working groups of the Internet Engineering Task Force, culminating in the publication of a series of RFCs, most notably RFC 2616, which defines HTTP/1.1, the version of HTTP in common use today.
HTTP is a request/response protocol between clients and servers. An HTTP client, such as a web browser, typically initiates a request by establishing a TCP connection to a particular port on a remote host (port 80 by default). An HTTP server listening on that port waits for the client to send a request string, such as "GET / HTTP/1.1" (which would request the default page of that web server), followed by an email-like MIME message which has a number of informational header strings that describe aspects of the request, followed by an optional body of arbitrary data. Some headers are optional, while others (such as Host) are required by the HTTP/1.1 protocol. Upon receiving the request, the server sends back a response string, such as "200 OK", and a message of its own, the body of which is perhaps the requested file, an error message, or some other information.
Resources used in the HTTP are identified using Uniform Resource Identifiers (URIs) in the http or https schemes.
In HTTP/0.9 and HTTP/1.0, a client sends a request to the server and then the server sends a response back to the client. After this, the connection is closed. HTTP/1.1, however, supports persistent connections. This enables the client to send a request and get a response, and then send additional requests and get additional responses. The TCP connection is not released for the multiple additional requests, so the relative overhead due to TCP is much less per request. The use of persistent connection is often called keep alive. It is also possible to send more than one (usually between two and five) request before getting responses from previous requests. This is called pipelining.
There is a HTTP/1.0 extension for connection persistence, but its utility is limited due to HTTP/1.0's lack of unambiguous message delimition rules. This extension uses a header called Keep-Alive, while the HTTP/1.1 connection persistence uses the Connection header. Therefore a HTTP/1.1 may choose to support either just HTTP/1.1 connection persistence, or both HTTP/1.0 and HTTP/1.1 connection persistence. Some HTTP/1.1 clients and servers do not implement connection persistence or have it disabled in their configuration.Both HTTP servers and clients are allowed to close TCP/IP connections at any time (i.e. depending on their settings, their load, etc.). This feature makes HTTP ideal for the World Wide Web, where pages regularly link to many other pages on the same server or to external servers. Closing an HTTP/1.1 connection can be a much longer operation (from 200 milliseconds up to several seconds) than closing an HTTP/1.0 connection, because the first usually needs a linger close while the second can be immediately closed as soon as the entire first request has been read and the full response has been sent.
HTTP can occasionally pose problems for Web developers (Web Applications), because HTTP is stateless (i.e. it does not keep session information) so this "feature" forces the use of alternative methods for maintaining users' "state". Many of these methods involve the use of cookies. HTTP is a URI scheme equivalent to the http scheme. It signals the browser to use HTTP with added encryption layer of SSL/TLS to protect the traffic. SSL is especially suited for HTTP since it can provide some protection even if only one side to the communication is authenticated. In the case in HTTP transactions over the Internet, typically only the server side is authenticated.
HTTP is a request/response protocol between clients and servers. An HTTP client, such as a web browser, typically initiates a request by establishing a TCP connection to a particular port on a remote host (port 80 by default). An HTTP server listening on that port waits for the client to send a request string, such as "GET / HTTP/1.1" (which would request the default page of that web server), followed by an email-like MIME message which has a number of informational header strings that describe aspects of the request, followed by an optional body of arbitrary data. Some headers are optional, while others (such as Host) are required by the HTTP/1.1 protocol. Upon receiving the request, the server sends back a response string, such as "200 OK", and a message of its own, the body of which is perhaps the requested file, an error message, or some other information.
Resources used in the HTTP are identified using Uniform Resource Identifiers (URIs) in the http or https schemes.
In HTTP/0.9 and HTTP/1.0, a client sends a request to the server and then the server sends a response back to the client. After this, the connection is closed. HTTP/1.1, however, supports persistent connections. This enables the client to send a request and get a response, and then send additional requests and get additional responses. The TCP connection is not released for the multiple additional requests, so the relative overhead due to TCP is much less per request. The use of persistent connection is often called keep alive. It is also possible to send more than one (usually between two and five) request before getting responses from previous requests. This is called pipelining.
There is a HTTP/1.0 extension for connection persistence, but its utility is limited due to HTTP/1.0's lack of unambiguous message delimition rules. This extension uses a header called Keep-Alive, while the HTTP/1.1 connection persistence uses the Connection header. Therefore a HTTP/1.1 may choose to support either just HTTP/1.1 connection persistence, or both HTTP/1.0 and HTTP/1.1 connection persistence. Some HTTP/1.1 clients and servers do not implement connection persistence or have it disabled in their configuration.Both HTTP servers and clients are allowed to close TCP/IP connections at any time (i.e. depending on their settings, their load, etc.). This feature makes HTTP ideal for the World Wide Web, where pages regularly link to many other pages on the same server or to external servers. Closing an HTTP/1.1 connection can be a much longer operation (from 200 milliseconds up to several seconds) than closing an HTTP/1.0 connection, because the first usually needs a linger close while the second can be immediately closed as soon as the entire first request has been read and the full response has been sent.
HTTP can occasionally pose problems for Web developers (Web Applications), because HTTP is stateless (i.e. it does not keep session information) so this "feature" forces the use of alternative methods for maintaining users' "state". Many of these methods involve the use of cookies. HTTP is a URI scheme equivalent to the http scheme. It signals the browser to use HTTP with added encryption layer of SSL/TLS to protect the traffic. SSL is especially suited for HTTP since it can provide some protection even if only one side to the communication is authenticated. In the case in HTTP transactions over the Internet, typically only the server side is authenticated.
FTP (File Transfer Protocol)
FTP or file transfer protocol is a protocol used for exchanging files over the Internet. FTP works in the same way as HTTP for transferring Web pages from a server to a user's browser, and SMTP for transferring electronic mail across the Internet in that FTP uses the Internet's TCP/IP protocols to enable data transfer. FTP is most commonly used to download a file from a server using the Internet or to upload a file to a server (e.g., uploading a Web page file to a server). While data is being transferred across the data stream, the control stream does not do anything. This can cause problems with large data transfers through a firewall, which will time out sessions after long periods of idleness. While the file may well be successfully transferred, the control session can be disconnected by the firewall, causing an error. FTP requires the user to login before data transfer can occur. However, anonymous access is also popular.
FTP is commonly run on two ports, 20 and 21. FTP can run over TCP as well as UDP, although TCP is much more common.
The FTP Server listens on Port 21 for incoming connection from FTP clients. A connection on this port forms the control stream, on which commands are passed to the FTP server.
For the actual file transfer to take place, a different connection is required. Depending on the transfer mode, the client (passive mode) or the server (active mode) can listen for the incoming data connection. Before file transfer begins, the client and server also negotiate the Port of the Data connection. In case of active connections, (where the server connects to the client to transfer data) the server binds on Port 20 before connecting to the client. For passive connections, there is no such restriction.
While data is being transferred via the data stream, the control stream sits idle. This can cause problems with large data transfers through firewalls which time out sessions after lengthy periods of idleness. While the file may well be successfully transferred, the control session can be disconnected by the firewall, causing an error to be generated.
The objectives of FTP, as outlined by its RFC, are to promote sharing of files (computer programs and/or data), to encourage indirect or implicit use of remote computers, to shield a user from variations in file storage systems among different hosts, to transfer data reliably and efficiently.
Disadvantages are passwords and file contents are sent in clear text in which can be intercepted by eavesdroppers, multiple TCP/IP connections are used in which one for the control connection and one for each download and upload, it is hard to filter active mode FTP traffic on the client side by using a firewall since the client must open an arbitrary port in order to receive the connection and this problem is largely resolved by using passive mode FTP, it is possible to abuse the protocol's built-in proxy features to tell a server to send data to an arbitrary port of a third computer, FTP is an extremely high latency protocol due to the number of commands needed to initiate a transfer, FTP is designed mainly for use by FTP client programs although is usable directly by a user at a terminal. Many sites that run FTP servers enable so-called "anonymous ftp". Under this arrangement, users do not need an account on the server. By default, the account name for the anonymous access is 'anonymous'. This account does not need a password. Users are commonly asked to send their email addresses as their passwords for authentication, usually there is trivial or no verification, depending on the FTP server and its configuration.
FTP is commonly run on two ports, 20 and 21. FTP can run over TCP as well as UDP, although TCP is much more common.
The FTP Server listens on Port 21 for incoming connection from FTP clients. A connection on this port forms the control stream, on which commands are passed to the FTP server.
For the actual file transfer to take place, a different connection is required. Depending on the transfer mode, the client (passive mode) or the server (active mode) can listen for the incoming data connection. Before file transfer begins, the client and server also negotiate the Port of the Data connection. In case of active connections, (where the server connects to the client to transfer data) the server binds on Port 20 before connecting to the client. For passive connections, there is no such restriction.
While data is being transferred via the data stream, the control stream sits idle. This can cause problems with large data transfers through firewalls which time out sessions after lengthy periods of idleness. While the file may well be successfully transferred, the control session can be disconnected by the firewall, causing an error to be generated.
The objectives of FTP, as outlined by its RFC, are to promote sharing of files (computer programs and/or data), to encourage indirect or implicit use of remote computers, to shield a user from variations in file storage systems among different hosts, to transfer data reliably and efficiently.
Disadvantages are passwords and file contents are sent in clear text in which can be intercepted by eavesdroppers, multiple TCP/IP connections are used in which one for the control connection and one for each download and upload, it is hard to filter active mode FTP traffic on the client side by using a firewall since the client must open an arbitrary port in order to receive the connection and this problem is largely resolved by using passive mode FTP, it is possible to abuse the protocol's built-in proxy features to tell a server to send data to an arbitrary port of a third computer, FTP is an extremely high latency protocol due to the number of commands needed to initiate a transfer, FTP is designed mainly for use by FTP client programs although is usable directly by a user at a terminal. Many sites that run FTP servers enable so-called "anonymous ftp". Under this arrangement, users do not need an account on the server. By default, the account name for the anonymous access is 'anonymous'. This account does not need a password. Users are commonly asked to send their email addresses as their passwords for authentication, usually there is trivial or no verification, depending on the FTP server and its configuration.
Hub
An Ethernet hub or concentrator is a device for connecting multiple twisted pair or fiber optic Ethernet devices together, making them act as a single segment. It works at the physical layer of the OSI model, repeating the signal that comes into one port out each of the other ports. If a signal comes into two ports at the same time a collision occurs, so every attached device shares the same collision domain. Hubs support only half duplex Ethernet, providing bandwidth which is shared among all the connected devices.
Most hubs detect typical problems such as excessive collisions on individual ports, and partition the port, disconnecting it from the shared medium. Thus hub-based Ethernet is generally more robust than coaxial cable based Ethernet where a misbehaving device can disable the entire segment. Even if not partitioned automatically, a hub makes troubleshooting easier because status lights can indicate the possible problem source or, as a last resort, devices can be disconnected from a hub one at a time much more easily than a coaxial cable.
Although switches are much more common, hubs are still useful in special circumstances:-A protocol analyzer connected to a switch does not always receive all the desired packets since the switch separates the ports into different segments. Connecting it to a hub allows it to see all the traffic going through the hub.
Some computer clusters require each member computer to receive all of the traffic going to the cluster. A hub will do this naturally; using a switch requires implementing special tricks.
When a switch is accessible for end users to make connections (for example, in a conference room), an inexperienced or careless user (or saboteur) can bring down the network by connecting two ports together, causing a loop. This can be prevented by using a hub, where a loop will break other users on the hub but not the rest of the network.
Most hubs detect typical problems such as excessive collisions on individual ports, and partition the port, disconnecting it from the shared medium. Thus hub-based Ethernet is generally more robust than coaxial cable based Ethernet where a misbehaving device can disable the entire segment. Even if not partitioned automatically, a hub makes troubleshooting easier because status lights can indicate the possible problem source or, as a last resort, devices can be disconnected from a hub one at a time much more easily than a coaxial cable.
Although switches are much more common, hubs are still useful in special circumstances:-A protocol analyzer connected to a switch does not always receive all the desired packets since the switch separates the ports into different segments. Connecting it to a hub allows it to see all the traffic going through the hub.
Some computer clusters require each member computer to receive all of the traffic going to the cluster. A hub will do this naturally; using a switch requires implementing special tricks.
When a switch is accessible for end users to make connections (for example, in a conference room), an inexperienced or careless user (or saboteur) can bring down the network by connecting two ports together, causing a loop. This can be prevented by using a hub, where a loop will break other users on the hub but not the rest of the network.
Switch
A switch is a device for making or breaking an electric circuit, or for selecting between multiple circuits. In the simplest case, a switch has two pieces of metal called contacts that touch to make a circuit, and separate to break the circuit. The contact material is chosen for its resistance to corrosion, because most metals form insulating oxides that would prevent the switch from working. Sometimes the contacts are plated with noble metals. They may be designed to wipe against each other to clean off any contamination. Nonmetallic conductors, such as conductive plastic, are sometimes used. The moving part that applies the operating force to the contacts is called the actuator, and may be a toggle or dolly, a rocker, a push-button or any type of mechanical linkage
A pair of contacts is said to be 'closed' when there is no space between them, allowing electricity to flow from one to the other. When the contacts are separated by a space, they are said to be 'open', and no electricity can flow. Switches can be classified according to the arrangement of their contacts. Some contacts are normally open until closed by operation of the switch, while others are normally closed and opened by the switch action. A switch with both types of contact is called a changeover switch. The terms pole and throw are used to describe switch contacts. A pole is a set of contacts that belong to a single circuit. A throw is one of two or more positions that the switch can adopt. These terms give rise to abbreviations for the types of switch which are used in the electronics industry. In mains wiring names generally involving the word way are used; however, these terms differ between British and American English and the terms two way and three way are used in both with different meanings. Switches with larger numbers of poles or throws can be described by replacing the "S" or "D" with a number or in some cases the letter T (for triple). In the rest of this article the terms SPST SPDT and intermediate will be used to avoid the ambiguity in the use of the word "way".
In a multi-throw switch, there are two possible transient behaviors as you move from one postion to another. In some switch designs, the new contact is made before the old contact is broken. This is known as make-before-break, and ensures that the moving contact never sees an open circuit. The alternative is break-before-make, where the old contact is broken before the new one is made. This ensures that the two contacts are never shorted to each other. Both types of design are in common use, for different applications
A biased switch is one containing a spring that returns the actuator to a certain position. The "on-off" notation can be modified by placing parentheses around all positions other than the resting position. For example, an (on)-off-(on) switch can be switched on by moving the actuator in either direction away from the centre, but returns to the central off position when the actuator is released.The momentary push-button switch is a type of biased switch. The most common type is a push-to-make switch, which makes contact when the button is pressed and breaks when the button is released. A push-to-break switch, on the other hand, breaks contact when the button is pressed and makes contact when it is released. An example of a push-to-break switch is a button used to release a door held open by an electromagnet. Changeover push button switches do exist but are even less common.
A pair of contacts is said to be 'closed' when there is no space between them, allowing electricity to flow from one to the other. When the contacts are separated by a space, they are said to be 'open', and no electricity can flow. Switches can be classified according to the arrangement of their contacts. Some contacts are normally open until closed by operation of the switch, while others are normally closed and opened by the switch action. A switch with both types of contact is called a changeover switch. The terms pole and throw are used to describe switch contacts. A pole is a set of contacts that belong to a single circuit. A throw is one of two or more positions that the switch can adopt. These terms give rise to abbreviations for the types of switch which are used in the electronics industry. In mains wiring names generally involving the word way are used; however, these terms differ between British and American English and the terms two way and three way are used in both with different meanings. Switches with larger numbers of poles or throws can be described by replacing the "S" or "D" with a number or in some cases the letter T (for triple). In the rest of this article the terms SPST SPDT and intermediate will be used to avoid the ambiguity in the use of the word "way".
In a multi-throw switch, there are two possible transient behaviors as you move from one postion to another. In some switch designs, the new contact is made before the old contact is broken. This is known as make-before-break, and ensures that the moving contact never sees an open circuit. The alternative is break-before-make, where the old contact is broken before the new one is made. This ensures that the two contacts are never shorted to each other. Both types of design are in common use, for different applications
A biased switch is one containing a spring that returns the actuator to a certain position. The "on-off" notation can be modified by placing parentheses around all positions other than the resting position. For example, an (on)-off-(on) switch can be switched on by moving the actuator in either direction away from the centre, but returns to the central off position when the actuator is released.The momentary push-button switch is a type of biased switch. The most common type is a push-to-make switch, which makes contact when the button is pressed and breaks when the button is released. A push-to-break switch, on the other hand, breaks contact when the button is pressed and makes contact when it is released. An example of a push-to-break switch is a button used to release a door held open by an electromagnet. Changeover push button switches do exist but are even less common.
Firewall
Today, firewalls are again using application level filters called proxies - or application level proxies because machines with modern CPU speeds are capable of doing deep inspection in reasonable time. These proxies can read the data part of each packet in order to make intelligent decisions about the connection. For example, http can be used to tunnel IRC or peer to peer file sharing protocols. Traditional stateful firewalls cannot detect this while an application level firewall can detect and selectively block http connections according to content.
Modern computers typically exchange data by breaking it up to network frames. These frames are called "packets" in TCP/IP, the most commonly used network protocol. Firewalls inspect each packet and decide whether it should be allowed to pass the firewall and continue travelling towards its destination, or discarded. Common ways of filtering packets are according to the source/destination address or according to the source/destination port.
But in most cases this information is not enough. The administrator of the firewall might want to allow packets to pass the firewall according to the context of the connection, and not just the individual packet characteristics. Therefore, a packet belonging to an existing connection, aimed at port 22 (the Secure Shell port) should be allowed to pass the firewall, but a packet that does not belong to any existing connection must be dropped.
With the traditional stateless firewalls, this was a problem, since the firewall had no way of knowing which packets belonged to existing connections and which didn't. Stateful firewalls solve this problem by monitoring network connections and matching any packets they inspect to existing or new connections. Therefore, they offer more fine grained control over network traffic.
Packet-filter Firewall- firewall can be used as a packet filter. It can forward or block packets based on the information in the network layer and transport layer headers: source and destination IP addresses, source and destination port addresses, and type of protocol (TCP or UDP). A packet-filter firewall is a router that uses a filtering table to decide which packets must be discarded (not forwarded).
Proxy Firewall- The packet-filter firewall is based on the information available in the network layer and transport layer headers (IP and TCP/UDP). However, sometimes we need to filter a message based on the information available in the message itself (at the application layer). As an example, assume that an organization wants to implement the following policies regarding its Web Pages: Only those Internet users who have previously established business relations with the company can have access; access to other users must be blocked. In this case, a packet-filter firewall is not feasible because it cannot distinguish between different packets arriving at TCP port 80 (HTTP). Testing must be done at the application level (using URLs). One solution is to install a proxy computer, which stands between the customer computer and the corporation computer.
Modern computers typically exchange data by breaking it up to network frames. These frames are called "packets" in TCP/IP, the most commonly used network protocol. Firewalls inspect each packet and decide whether it should be allowed to pass the firewall and continue travelling towards its destination, or discarded. Common ways of filtering packets are according to the source/destination address or according to the source/destination port.
But in most cases this information is not enough. The administrator of the firewall might want to allow packets to pass the firewall according to the context of the connection, and not just the individual packet characteristics. Therefore, a packet belonging to an existing connection, aimed at port 22 (the Secure Shell port) should be allowed to pass the firewall, but a packet that does not belong to any existing connection must be dropped.
With the traditional stateless firewalls, this was a problem, since the firewall had no way of knowing which packets belonged to existing connections and which didn't. Stateful firewalls solve this problem by monitoring network connections and matching any packets they inspect to existing or new connections. Therefore, they offer more fine grained control over network traffic.
Packet-filter Firewall- firewall can be used as a packet filter. It can forward or block packets based on the information in the network layer and transport layer headers: source and destination IP addresses, source and destination port addresses, and type of protocol (TCP or UDP). A packet-filter firewall is a router that uses a filtering table to decide which packets must be discarded (not forwarded).
Proxy Firewall- The packet-filter firewall is based on the information available in the network layer and transport layer headers (IP and TCP/UDP). However, sometimes we need to filter a message based on the information available in the message itself (at the application layer). As an example, assume that an organization wants to implement the following policies regarding its Web Pages: Only those Internet users who have previously established business relations with the company can have access; access to other users must be blocked. In this case, a packet-filter firewall is not feasible because it cannot distinguish between different packets arriving at TCP port 80 (HTTP). Testing must be done at the application level (using URLs). One solution is to install a proxy computer, which stands between the customer computer and the corporation computer.
BOOTP
In computing, BOOTP, short for Bootstrap Protocol, is a UDP network protocol used by a network client to obtain its IP address automatically. This is usually done in the bootstrap process of computers or operating systems running on them. The BOOTP servers assign the IP address from a pool of addresses to each client. The protocol was originally defined in RFC 951.
BOOTP enables 'diskless workstation' computers to obtain an IP address prior to loading any advanced operating system. Historically, it has been used for Unix-like diskless workstations which also obtained the location of their boot image using this protocol and also by corporations to roll out a pre-configured client installation to newly purchased PCs.
Originally requiring the use of a boot floppy disk to establish the initial network connection, the protocol became embedded in the BIOS of some network cards themselves and in many modern motherboards thus allowing direct network booting.
Recently those with an interest in diskless stand-alone media center PCs have shown new interest in this method of booting a Windows operating system. DHCP is a more advanced protocol based on BOOTP, but is far more complex to implement. Most DHCP servers also offer BOOTP support.
BOOTP enables 'diskless workstation' computers to obtain an IP address prior to loading any advanced operating system. Historically, it has been used for Unix-like diskless workstations which also obtained the location of their boot image using this protocol and also by corporations to roll out a pre-configured client installation to newly purchased PCs.
Originally requiring the use of a boot floppy disk to establish the initial network connection, the protocol became embedded in the BIOS of some network cards themselves and in many modern motherboards thus allowing direct network booting.
Recently those with an interest in diskless stand-alone media center PCs have shown new interest in this method of booting a Windows operating system. DHCP is a more advanced protocol based on BOOTP, but is far more complex to implement. Most DHCP servers also offer BOOTP support.
DHCP
Dynamic Host Configuration Protocol (DHCP) is a client-server networking protocol. A DHCP server provides configuration parameters specific to the DHCP client host requesting, generally, information required by the client host to participate on an IP network. DHCP also provides a mechanism for allocation of IP addresses to client hosts. DHCP appeared as a standard protocol in October 1993. RFC 2131 provides the latest (March 1997) DHCP definition. The latest standard on a protocol describing DHCPv6, DHCP in a IPv6 environment, was published in July 2003 as RFC 3315
The DHCP protocol provides three methods of IP-address allocation:
Manual Allocation, where the DHCP server performs the allocation based on a table with MAC address - IP address pairs manually filled by the server administrator. Only requesting clients with a MAC address listed in this table get the IP address according to the table. Automatic Allocation, where the DHCP server permanently assigns to a requesting client a free IP-address from a range given by the administrator. Dynamic Allocation, the only method which provides dynamic re-use of IP addresses. A network administrator assigns a range of IP addresses to DHCP, and each client computer on the LAN has its TCP/IP software configured to request an IP address from the DHCP server when that client computer's network interface card starts up. The request-and-grant process uses a lease concept with a controllable time period. This eases the network installation procedure on the client computer side considerably. Some DHCP server implementations can update the DNS name associated with the client hosts to reflect the new IP address. They make use of the DNS update protocol established with RFC 2136
DHCP is used by most cable internet in the U.S. to allocate IP addresses. DSL providers in the US rarely use DHCP, using PPPoE instead. In addition, several routers provide DHCP support for networks of up to 255 computers, for assigning private IP addresses.
Microsoft introduced DHCP on their NT server with Windows NT version 3.5 in late 1994. Despite being called "a new feature from Microsoft", DHCP did not originate from Microsoft. The Internet Software Consortium published DHCP software distributions for Unix variants with version 1.0.0 of the ISC DHCP Server released on December 6, 1997 and a more RFC-compliant version 2.0 on June 22, 1999. One can download this software from http://www.isc.org/sw/dhcp/Novell has included a DHCP server in their NetWare operating system since version 5, released in 1998. It integrates with Novell's directory service - Novell eDirectory. Other major implementations include:
Cisco with a DHCP server made available in Cisco IOS 12.0 in February 1999 Sun, who added DHCP support in the July 2001 release of Solaris 8. Cisco Systems offers DHCP servers in routers and switches with their IOS software. Moreover, they offer Cisco Network Registrar (CNR) - a highly scalable and flexible DNS, DHCP and TFTP server.
The DHCP protocol provides three methods of IP-address allocation:
Manual Allocation, where the DHCP server performs the allocation based on a table with MAC address - IP address pairs manually filled by the server administrator. Only requesting clients with a MAC address listed in this table get the IP address according to the table. Automatic Allocation, where the DHCP server permanently assigns to a requesting client a free IP-address from a range given by the administrator. Dynamic Allocation, the only method which provides dynamic re-use of IP addresses. A network administrator assigns a range of IP addresses to DHCP, and each client computer on the LAN has its TCP/IP software configured to request an IP address from the DHCP server when that client computer's network interface card starts up. The request-and-grant process uses a lease concept with a controllable time period. This eases the network installation procedure on the client computer side considerably. Some DHCP server implementations can update the DNS name associated with the client hosts to reflect the new IP address. They make use of the DNS update protocol established with RFC 2136
DHCP is used by most cable internet in the U.S. to allocate IP addresses. DSL providers in the US rarely use DHCP, using PPPoE instead. In addition, several routers provide DHCP support for networks of up to 255 computers, for assigning private IP addresses.
Microsoft introduced DHCP on their NT server with Windows NT version 3.5 in late 1994. Despite being called "a new feature from Microsoft", DHCP did not originate from Microsoft. The Internet Software Consortium published DHCP software distributions for Unix variants with version 1.0.0 of the ISC DHCP Server released on December 6, 1997 and a more RFC-compliant version 2.0 on June 22, 1999. One can download this software from http://www.isc.org/sw/dhcp/Novell has included a DHCP server in their NetWare operating system since version 5, released in 1998. It integrates with Novell's directory service - Novell eDirectory. Other major implementations include:
Cisco with a DHCP server made available in Cisco IOS 12.0 in February 1999 Sun, who added DHCP support in the July 2001 release of Solaris 8. Cisco Systems offers DHCP servers in routers and switches with their IOS software. Moreover, they offer Cisco Network Registrar (CNR) - a highly scalable and flexible DNS, DHCP and TFTP server.
SNMP
The Simple Network Management Protocol (SNMP) forms part of the internet protocol suite as defined by the Internet Engineering Task Force. The protocol can support monitoring of network-attached devices for any conditions that warrant administrative attention. The SNMP protocol is extensible by design. This is achieved through the notion of a management information base or MIB, which specifies the management data of a specific subsystem of an SNMP-enabled device, using a hierarchical namespace containing object identifiers, implemented via ASN.1. The MIB hierarchy can be depicted as a tree with a nameless root, the levels of which are assigned by different organizations. This model permits management across all layers of the OSI reference model, extending into applications such as databases, email, and the J2EE reference model, as MIBs can be defined for all such area-specific information and operations.
Architecturally, the SNMP framework has three fundamental components: Master Agents, Subagents and Management Stations.
A master agent is a piece of software running on an SNMP-capable network component (say, a router). that responds to SNMP requests made by a management station. Thus it acts as a server in client-server architecture terminology or as a daemon in operating system terminology. A master agent relies on subagents to provide information about or management of specific functionality. Master agent can also be referred as Managed objects. A subagent is a piece of software running on an SNMP-capable network component that implements the information and management functionality defined by a specific MIB / of a specific subsystem like, for example, the ethernet link layer. Some capabilities of the subagent are gathering of information from the managed objects, configuring parameters of the managed object, responding to manager's request and generates alarm, or rather called traps to managers. The manager or management station is the final component in the architecture of SNMP. It functions as the equivalent of a client in a client-server architecture. It issues requests for management operations on behalf of an administrator or application, and receives traps from agents as well.
The SNMP protocol operates at the application layer (layer 7) of the OSI model. It specified (in version 1) five core protocol data units (PDUs):
Normally, a network management system is able to manage device with SNMP agent installed. However in the absence of the SNMP agent, it can be managed with the help of a proxy agent. The SNMP agent associated with the proxy policy is called a proxy agent, or commercially a proxy server. The proxy agent monitor non-SNMP Community with non-SNMP agents and then converts the objects and data to SNMP compatible objects and data tobe fed to an SNMP manager.
Architecturally, the SNMP framework has three fundamental components: Master Agents, Subagents and Management Stations.
A master agent is a piece of software running on an SNMP-capable network component (say, a router). that responds to SNMP requests made by a management station. Thus it acts as a server in client-server architecture terminology or as a daemon in operating system terminology. A master agent relies on subagents to provide information about or management of specific functionality. Master agent can also be referred as Managed objects. A subagent is a piece of software running on an SNMP-capable network component that implements the information and management functionality defined by a specific MIB / of a specific subsystem like, for example, the ethernet link layer. Some capabilities of the subagent are gathering of information from the managed objects, configuring parameters of the managed object, responding to manager's request and generates alarm, or rather called traps to managers. The manager or management station is the final component in the architecture of SNMP. It functions as the equivalent of a client in a client-server architecture. It issues requests for management operations on behalf of an administrator or application, and receives traps from agents as well.
The SNMP protocol operates at the application layer (layer 7) of the OSI model. It specified (in version 1) five core protocol data units (PDUs):
Normally, a network management system is able to manage device with SNMP agent installed. However in the absence of the SNMP agent, it can be managed with the help of a proxy agent. The SNMP agent associated with the proxy policy is called a proxy agent, or commercially a proxy server. The proxy agent monitor non-SNMP Community with non-SNMP agents and then converts the objects and data to SNMP compatible objects and data tobe fed to an SNMP manager.
IPv4 & IPv6
IPv6 was recommended by the IPv6 Area Directors of the Internet Engineering Task Force at the Toronto IETF meeting on July 25, 1994, and documented in RFC 1752, "The Recommendation for the IP Next Generation Protocol". The recommendation was approved by the Internet Engineering Steering Group on November 17, 1994 and made a Proposed Standard.
The current version of the Internet Protocol is version 4 referred to as IPv4. IPv6 is a new version of IP which is designed to be an evolutionary step from IPv4. It is a natural increment to IPv4. It can be installed as a normal software upgrade in internet devices and is interoperable with the current IPv4. Its deployment strategy was designed to not have any "flag" days. IPv6 is designed to run well on high performance networks such as ATM and at the same time is still efficient for low bandwidth networks such as wireless. In addition, it provides a platform for new internet functionality that will be required in the near future. Pv6 was designed to take an evolutionary step from IPv4. It was not a design goal to take a radical step away from IPv4. Functions which work in IPv4 were kept in IPv6. Functions which didn't work were removed. The changes from IPv4 to IPv6 fall primarily into the following categories:
Expanded Routing and Addressing Capabilities IPv6 increases the IP address size from 32 bits to 128 bits, to support more levels of addressing hierarchy and a much greater number of addressable nodes, and simpler auto-configuration of addresses. The scalability of multicast routing is improved by adding a "scope" field to multicast addresses.
A new type of address called a "anycast address" is defined, to identify sets of nodes where a packet sent to an anycast address is delivered to one of the nodes. The use of anycast addresses in the IPv6 source route allows nodes to control the path which their traffic flows.
Header Format Simplification Some IPv4 header fields have been dropped or made optional, to reduce the common-case processing cost of packet handling and to keep the bandwidth cost of the IPv6 header as low as possible despite the increased size of the addresses. Even though the IPv6 addresses are four time longer than the IPv4 addresses, the IPv6 header is only twice the size of the IPv4 header.
Improved Support for Options Changes in the way IP header options are encoded allows for more efficient forwarding, less stringent limits on the length of options, and greater flexibility for introducing new options in the future.
Quality-of-Service Capabilities A new capability is added to enable the labeling of packets belonging to particular traffic "flows" for which the sender requests special handling, such as non-default quality of service or "real- time" service.
Authentication and Privacy Capabilities IPv6 includes the definition of extensions which provide support for authentication, data integrity, and confidentiality. This is included as a basic element of IPv6 and will be included in all implementations.
The IPv6 protocol consists of two parts, the basic IPv6 header and IPv6 extension headers.
There are a number of reasons why IPv6 is appropriate for the next generation of the Internet Protocol. It solves the Internet scaling problem, provides a flexible transition mechanism for the current Internet, and was designed to meet the needs of new markets such as nomadic personal computing devices, networked entertainment, and device control. It does this in a evolutionary way which reduces the risk of architectural problems.
Ease of transition is a key point in the design of IPv6. It is not something was added in at the end. IPv6 is designed to interoperate with IPv4. Specific mechanisms were built into IPv6 to support transition and compatibility with IPv4. It was designed to permit a gradual and piecemeal deployment with a minimum of dependencies.
IPv6 supports large hierarchical addresses which will allow the Internet to continue to grow and provide new routing capabilities not built into IPv4. It has anycast addresses which can be used for policy route selection and has scoped multicast addresses which provide improved scalability over IPv4 multicast. It also has local use address mechanisms which provide the ability for "plug and play" installation.
The address structure of IPv6 was also designed to support carrying the addresses of other internet protocol suites. Space was allocated in the addressing plan for IPX and NSAP addresses. This was done to facilitate migration of these internet protocols to IPv6.
IPv6 provides a platform for new Internet functionality. This includes support for real-time flows, provider selection, host mobility, end-to- end security, auto-configuration, and auto-reconfiguration.
In summary, IPv6 is a new version of IP. It can be installed as a normal software upgrade in internet devices. It is interoperable with the current IPv4. Its deployment strategy was designed to not have any "flag" days. IPv6 is designed to run well on high performance networks such as ATM and at the same time is still efficient for low bandwidth networks such as wireless. In addition, it provides a platform for new internet functionality that will be required in the near future.
The current version of the Internet Protocol is version 4 referred to as IPv4. IPv6 is a new version of IP which is designed to be an evolutionary step from IPv4. It is a natural increment to IPv4. It can be installed as a normal software upgrade in internet devices and is interoperable with the current IPv4. Its deployment strategy was designed to not have any "flag" days. IPv6 is designed to run well on high performance networks such as ATM and at the same time is still efficient for low bandwidth networks such as wireless. In addition, it provides a platform for new internet functionality that will be required in the near future. Pv6 was designed to take an evolutionary step from IPv4. It was not a design goal to take a radical step away from IPv4. Functions which work in IPv4 were kept in IPv6. Functions which didn't work were removed. The changes from IPv4 to IPv6 fall primarily into the following categories:
Expanded Routing and Addressing Capabilities IPv6 increases the IP address size from 32 bits to 128 bits, to support more levels of addressing hierarchy and a much greater number of addressable nodes, and simpler auto-configuration of addresses. The scalability of multicast routing is improved by adding a "scope" field to multicast addresses.
A new type of address called a "anycast address" is defined, to identify sets of nodes where a packet sent to an anycast address is delivered to one of the nodes. The use of anycast addresses in the IPv6 source route allows nodes to control the path which their traffic flows.
Header Format Simplification Some IPv4 header fields have been dropped or made optional, to reduce the common-case processing cost of packet handling and to keep the bandwidth cost of the IPv6 header as low as possible despite the increased size of the addresses. Even though the IPv6 addresses are four time longer than the IPv4 addresses, the IPv6 header is only twice the size of the IPv4 header.
Improved Support for Options Changes in the way IP header options are encoded allows for more efficient forwarding, less stringent limits on the length of options, and greater flexibility for introducing new options in the future.
Quality-of-Service Capabilities A new capability is added to enable the labeling of packets belonging to particular traffic "flows" for which the sender requests special handling, such as non-default quality of service or "real- time" service.
Authentication and Privacy Capabilities IPv6 includes the definition of extensions which provide support for authentication, data integrity, and confidentiality. This is included as a basic element of IPv6 and will be included in all implementations.
The IPv6 protocol consists of two parts, the basic IPv6 header and IPv6 extension headers.
There are a number of reasons why IPv6 is appropriate for the next generation of the Internet Protocol. It solves the Internet scaling problem, provides a flexible transition mechanism for the current Internet, and was designed to meet the needs of new markets such as nomadic personal computing devices, networked entertainment, and device control. It does this in a evolutionary way which reduces the risk of architectural problems.
Ease of transition is a key point in the design of IPv6. It is not something was added in at the end. IPv6 is designed to interoperate with IPv4. Specific mechanisms were built into IPv6 to support transition and compatibility with IPv4. It was designed to permit a gradual and piecemeal deployment with a minimum of dependencies.
IPv6 supports large hierarchical addresses which will allow the Internet to continue to grow and provide new routing capabilities not built into IPv4. It has anycast addresses which can be used for policy route selection and has scoped multicast addresses which provide improved scalability over IPv4 multicast. It also has local use address mechanisms which provide the ability for "plug and play" installation.
The address structure of IPv6 was also designed to support carrying the addresses of other internet protocol suites. Space was allocated in the addressing plan for IPX and NSAP addresses. This was done to facilitate migration of these internet protocols to IPv6.
IPv6 provides a platform for new Internet functionality. This includes support for real-time flows, provider selection, host mobility, end-to- end security, auto-configuration, and auto-reconfiguration.
In summary, IPv6 is a new version of IP. It can be installed as a normal software upgrade in internet devices. It is interoperable with the current IPv4. Its deployment strategy was designed to not have any "flag" days. IPv6 is designed to run well on high performance networks such as ATM and at the same time is still efficient for low bandwidth networks such as wireless. In addition, it provides a platform for new internet functionality that will be required in the near future.
7 LAYERS OF OSI
OSI short for Open System Interconnection, an ISO standard for worldwide communications that defines a networking framework for implementing protocols in seven layers. ISO shorts for International Organization for Standardization. Founded in 1946, ISO is an international organization composed of national standards bodies from over 75 countries. For example, ANSI (American National Standards Institute) is a member of ISO. ISO has defined a number of important computer standards; the most significant of which is perhaps OSI (Open Systems Interconnection) a standardized architecture for designing networks.
In OSI layers control is passed from one layer to the next, starting at the application layer in one station, proceeding to the bottom layer, over the channel to the next station and back up the hierarchy.
In communications the term channel refers to a communications path between two computers or devices. It can refer to the physical medium (the wires) or to a set of properties that distinguishes one channel from another. For example, TV channels refer to particular frequencies at which radio waves are transmitted. IRC channels refer to specific discussions.
Most of the functionality in the OSI model exists in all communications systems, although two or three OSI layers may be incorporated into one.
OSI is also referred to as the OSI Reference Model or just the OSI Model.
Physical Layer
The physical later is concerned with transmitting raw bits over a communication channel. The design issues have to do with making sure that when one side sends a 1 bit, the other side as a 1 bit, not as a 0 bit receives it. Typical questions here are how many volts should be used to represent a 1 and how many for a 0, how many microseconds a bit lasts, whether transmission may proceed simultaneously in both directions, how the initial connection is established and how it is torn down when both sides are finished, and how many pins the network connector has and what each pin is used for. The design issues here deal largely with mechanical, electrical, and procedural interfaces, and the physical transmission medium, which lies below the physical layer. Physical layer design can properly be considered to be within the domain of the electrical engineer.
Data Link Layer
The main task of the data link layer is to take a raw transmission facility and transform it into a line that appears free of transmission errors in the network layer. It accomplishes this task by having the sender break the input data up into data frames (typically a few hundred bytes), transmit the frames sequentially, and process the acknowledgment frames sent back by the receiver. Since the physical layer merely accepts and transmits a stream of bits without any regard to meaning of structure, it is up to the data link layer to create and recognize frame boundaries. This can be accomplished by attaching special bit patterns to the beginning and end of the frame. If there is a chance that these bit patterns might occur in the data, special care must be taken to avoid confusion. The data link layer should provide error control between adjacent nodes.
A noise burst on the line can destroy a frame completely. In this case, the data link layer software on the source machine must retransmit the frame. However, multiple transmissions of the same frame introduce the possibility of duplicate frames. A duplicate frame could be sent, for example, if the acknowledgment frame from the receiver back to the sender was destroyed. It is up to this layer to solve the problems caused by damaged, list, and duplicate frames. The data link layer may offer several different service classes to the network layer, each of a different quality and with a different price.
Another issue that arises in the data link layer (and most of the higher layers as well) is how to keep a fast transmitter from drowning a slow receiver in data. Some traffic regulation mechanism must be employed in order to let the transmitter know how much buffer space the receiver has at the moment. Frequently, flow regulation and error handling are integrated, for convenience.
If the line can be used to transmit data in both directions, this introduces a new complication that the data link layer software must deal with. The problem is that the acknowledgment frames for A to B traffic competes for the use of the line with data frames for the B to A traffic. A clever solution piggybacking has been devised.
In most practical situations, there is a need for transmitting data in both directions. One way of achieving full-duplex data transmission would be to have two separate communication channels, and use each one for simplex data traffic (in different directions). If this were done, we would have two separate physical circuits, each with a "forward" channel (for data) and a "reverse" channel (for acknowledgment). In both cases the bandwidth of the reverse channel would be almost entirely wasted. In effect, the user would be paying the cost of two circuits but only using the capacity of one.
A better idea is to use the same circuit for data in both directions. In this model the data frames from A to B are intermixed with the acknowledgment frames from A to B. By looking at the "kind" field in the header of an incoming frame, the receiver can tell whether the frame is data or acknowledgment.
Although interweaving data and control frames on the same circuit is an improvement over having two separate physical circuits, yet another improvement is possible. When a data frame arrives, instead of immediately sending a separate control frame, the receiver restrains it and waits until the network layer passes it the next packet. The acknowledgment is attached to the outgoing data frame. In effect, the acknowledgment gets a free ride on the next outgoing data frame. The technique of temporarily delaying outgoing acknowledgment so that they can be hooked onto the next outgoing data frame is widely known as piggybacking.
Network Layer
This layer provides switching and routing technologies, creating logical paths, known as virtual circuits for transmitting data from node. Routing and forwarding are functions of this layer, as well as addressing, internetworking error handling, congestion control and packet sequencing.
The network layer is concerned with controlling the operation of the subnet. A key design issue is determining how packets are routed from source to destination. Routes could be based on static tables that are "wired into" the network and rarely changed. They could also be determined at the start of each conversation, for example a terminal session. Finally, they could be highly dynamic, being determined anew for each packet, to reflect the current network load.
If too many packets are present in the subnet at the same time, they will get in each other's way, forming bottlenecks. The control of such congestion also belongs to the network layer.
Since the operators of the subnet may well expect remuneration for their efforts, there is often some accounting function built into the network layer. At the very least, the software must count how many packets or characters or each customer sends bits, to produce billing information. When a packet crosses a national border, with different rates on each side, the accounting can become complicated.
When a packet has to travel from one network to another to get to its destination, many problems can arise. The addressing used by the second network may be different from the first one. The second one may not accept the packet at all because it is too large. The protocols may differ, and so on. It is up to the network layer to overcome all these problems to allow heterogeneous networks to be interconnected. In broadcast networks, the routing problem is simple, so the network layer is often thin or even nonexistent.
NFS uses Internetwork Protocol (IP) as its network layer interface. IP is responsible for routing, directing datagrams from one network to another. The network layer may have to break large datagrams, larger than MTU, into smaller packets and host receiving the packet will have to reassemble the fragmented datagram. The Internetwork Protocol identifies each host with a 32-bit IP address. IP addresses are written as four dot-separated decimal numbers between 0 and 255, e.g., 129.79.16.40. The leading 1-3 bytes of the IP identify the network and the remaining bytes identify the host on that network. The network portion of the IP is assigned by InterNIC Registration Services, under the contract to the National Science Foundation, and the local network administrators assign the host portion of the IP, locally by noc@indiana.edu. For large sites, usually subnetted like ours, the first two bytes represent the network portion of the IP, and the third and fourth bytes identify the subnet and host respectively. Even though IP packets are addressed using IP addresses, hardware addresses must be used to actually transport data from one host to another. The Address Resolution Protocol (ARP) is used to map the IP address to it hardware.
Transport Layer
This layer provides transparent transfer of data between end systems, or hosts, and is responsible for end-to-end error recovery and flow control. It ensures complete data transfer.
The basic function of the transport layer is to accept data from the session layer, split it up into smaller units if need be, pass these to the network layer, and ensure that the pieces all arrive correctly at the other end. Furthermore, all this must be done efficiently, and in a way that isolates the session layer from the inevitable changes in the hardware technology.
Under normal conditions, the transport layer creates a distinct network connection for each transport connection required by the session layer. If the transport connection requires a high throughput, however, the transport layer might create multiple network connections, dividing the data among the network connections to improve throughput. On the other hand, if creating or maintaining a network connection is expensive, the transport layer might multiplex several transport connections onto the same network connection to reduce the cost. In all cases, the transport layer is required to make the multiplexing transparent to the session layer.
The transport layer also determines what type of service to provide to the session layer, and ultimately, the users of the network. The most popular type of transport connection is an error-free point-to-point channel that delivers messages in the order in which they were sent. However, other possible kinds of transport, service and transport isolated messages with no guarantee about the order of delivery, and broadcasting of messages to multiple destinations. The type of service is determined when the connection is established.
The transport layer is a true source-to-destination or end-to-end layer. In other words, a program on the source machine carries on a conversation with a similar program on the destination machine, using the message headers and control messages.
Many hosts are multi-programmed, which implies that multiple connections will be entering and leaving each host. Their needs to be some way to tell which message belong to which connection. The transport header is one place this information could be put.
In addition to multiplexing several message streams onto one channel, the transport layer musk takes care of establishing and deleting connections across the network. This requires some kind of naming mechanism, so that process on one machine has a way of describing with whom it wishes to converse. There must also be a mechanism to regulate the flow of information, so that a fast host cannot overrun a slow one. Flow control between hosts is distinct from flow control between switches, although similar principles apply to both.
Session Layer
This layer establishes, manages and terminates connections between applications. The session layer sets up, coordinates, and terminates conversations, exchanges, and dialogues between the applications at each end. It deals with session and connection coordination.
The session layer allows users on different machines to establish sessions between them. A session allows ordinary data transport, as does the transport layer, but it also provides some enhanced services useful in some applications. A session might be used to allow a user to log into a remote time-sharing system or to transfer a file between two machines.
One of the services of the session layer is to manage dialogue control. Sessions can allow traffic to go in both directions at the same time, or in only one direction at a time. If traffic can only go one way at a time, the session layer can help keep track of whose turn it is.
A related session service is token management. For some protocols, it is essential that both sides do not attempt the same operation at the same time. To manage these activities, the session layer provides tokens that can be exchanged. Only the side holding the token may perform the critical operation.
Another session service is synchronization. Consider the problems that might occur when trying to do a two-hour file transfer between two machines on a network with a 1-hour mean time between crashes. After each transfer was aborted, the whole transfer would have to start over again, and would probably fail again with the next network crash. To eliminate this problem, the session layer provides a way to insert checkpoints into the data stream, so that after a crash, only the data after the last checkpoint has to be repeated.
Presentation Layer
This layer provides independence from differences in data representation (e.g., encryption by translating from application to network format, and vice versa. The presentation layer works to transform data into the form that the application layer can accept. This layer formats and encrypts data to be sent across a network, providing freedom from compatibility problems. It is sometimes called the syntax layer.
The presentation layer performs certain functions that are requested sufficiently often to warrant finding a general solution for them, rather than letting each user solve the problems. In particular, unlike all the lower layers, which are just interested in moving bits reliably from here to there, the presentation layer is concerned with the syntax and semantics of the information transmitted.
A typical example of a presentation service is encoding data in a standard, agreed upon way. Most user programs do not exchange random binary bit strings. They exchange things such as people's names, dates, amounts of money, and invoices. These items are represented as character strings, integers, floating point numbers, and data structures composed of several simpler items.
Different computers have different codes for representing character strings, integers and so on. In order to make it possible for computers with different representation to communicate, the data structures to be exchanged can be defined in an abstract way, along with a standard encoding to be used "on the wire". The presentation layer handles the job of managing these abstract data structures and converting from the representation used inside the computer to the network standard representation.
The presentation layer is also concerned with other aspects of information representation. For example, data compression can be used here to reduce the number of bits that have to be transmitted and cryptography is frequently required for privacy and authentication.
Application Layer
This layer supports application and end-user processes. Communication partners are identified, quality of service is identified, user authentication and privacy are considered, and any constraints on data syntax are identified. Everything at this layer is application-specific. This layer provides application services for file transfers, e-mail and other network software services. Telnet and FTP are applications that exist entirely in the application level. Tiered application architectures are part of this layer.
The application layer contains a variety of protocols that are commonly needed. For example, there are hundreds of incompatible terminal types in the world. Consider the plight of a full screen editor that is supposed to work over a network with many different terminal types, each with different screen layouts, escape sequences for inserting and deleting text, moving the cursor, etc.
One way to solve this problem is to define an abstract network virtual terminal for which editors and other programs can be written to deal with. To handle each terminal type, a piece of software must be written to map the functions of the network virtual terminal onto the real terminal. For example, when the editor moves the virtual terminal's cursor to the upper left-hand corner of the screen, this software must issue the proper command sequence to the real terminal to get its cursor there too. All the virtual terminal software is in the application layer.
Another application layer function is file transfer. Different file systems have different file naming conventions, different ways of representing text lines, and so on. Transferring a file between two different systems requires handling these and other incompatibilities. This work, too, belongs to the application layer, as do electronic mail, remote job entry, directory lookup, and various other general-purpose and special-purpose facilities.
Network Architectures:
Ethernet
Ethernet is the most popular physical layer LAN technology in use today. Other LAN types include Token Ring, Fast Ethernet, Fiber Distributed Data Interface (FDDI), Asynchronous Transfer Mode (ATM) and LocalTalk. Ethernet is popular because it strikes a good balance between speed, cost and ease of installation. These benefits, combined with wide acceptance in the computer marketplace and the ability to support virtually all popular network protocols, make Ethernet an ideal networking technology for most computer users today. The Institute for Electrical and Electronic Engineers (IEEE) defines the Ethernet standard as IEEE Standard 802.3. This standard defines rules for configuring an Ethernet network as well as specifying how elements in an Ethernet network interact with one another. By adhering to the IEEE standard, network equipment and network protocols can communicate efficiently.
Fast Ethernet
For Ethernet networks that need higher transmission speeds, the Fast Ethernet standard (IEEE 802.3u) has been established. This standard raises the Ethernet speed limit from 10 Megabits per second (Mbps) to 100 Mbps with only minimal changes to the existing cable structure. There are three types of Fast Ethernet: 100BASE-TX for use with level 5 UTP cable, 100BASE-FX for use with fiber-optic cable, and 100BASE-T4 which utilizes an extra two wires for use with level 3 UTP cable. The 100BASE-TX standard has become the most popular due to its close compatibility with the 10BASE-T Ethernet standard. For the network manager, the incorporation of Fast Ethernet into an existing configuration presents a host of decisions. Managers must determine the number of users in each site on the network that need the higher throughput, decide which segments of the backbone need to be reconfigured specifically for 100BASE-T and then choose the necessary hardware to connect the 100BASE-T segments with existing 10BASE-T segments. Gigabit Ethernet is a future technology that promises a migration path beyond Fast Ethernet so the next generation of networks will support even higher data transfer speeds.
Token Ring
Token Ring is another form of network configuration which differs from Ethernet in that all messages are transferred in a unidirectional manner along the ring at all times. Data is transmitted in tokens, which are passed along the ring and viewed by each device. When a device sees a message addressed to it, that device copies the message and then marks that message as being read. As the message makes its way along the ring, it eventually gets back to the sender who now notes that the message was received by the intended device. The sender can then remove the message and free that token for use by others.
Various PC vendors have been proponents of Token Ring networks at different times and thus these types of networks have been implemented in many organizations.
Various PC vendors have been proponents of Token Ring networks at different times and thus these types of networks have been implemented in many organizations.
FDDI
FDDI (Fiber-Distributed Data Interface) is a standard for data transmission on fiber optic lines in a local area network that can extend in range up to 200 km (124 miles). The FDDI protocol is based on the token ring protocol. In addition to being large geographically, an FDDI local area network can support thousands of users.
Protocols:
Network protocols are standards that allow computers to communicate. A protocol defines how computers identify one another on a network, the form that the data should take in transit, and how this information is processed once it reaches its final destination. Protocols also define procedures for handling lost or damaged transmissions or "packets." TCP/IP (for UNIX, Windows NT, Windows 95 and other platforms), IPX (for Novell NetWare), DECnet (for networking Digital Equipment Corp. computers), AppleTalk (for Macintosh computers), and NetBIOS/NetBEUI (for LAN Manager and Windows NT networks) are the main types of network protocols in use today.
Although each network protocol is different, they all share the same physical cabling. This common method of accessing the physical network allows multiple protocols to peacefully coexist over the network media, and allows the builder of a network to use common hardware for a variety of protocols. This concept is known as "protocol independence,"
Some Important Protocols and their job:
Although each network protocol is different, they all share the same physical cabling. This common method of accessing the physical network allows multiple protocols to peacefully coexist over the network media, and allows the builder of a network to use common hardware for a variety of protocols. This concept is known as "protocol independence,"
Some Important Protocols and their job:
Protocol | Acronym | Its Job |
Point-To-Point | TCP/IP | The backbone protocol of the internet. Popular also for intranets using the internet |
Transmission Control Protocol/internet Protocol | TCP/IP | The backbone protocol of the internet. Popular also for intranets using the internet |
Internetwork Package Exchange/Sequenced Packet Exchange | IPX/SPX | This is a standard protocol for Novell Network Operating System |
NetBIOS Extended User Interface | NetBEUI | This is a Microsoft protocol that doesn't support routing to other networks |
File Transfer Protocol | FTP | Used to send and receive files from a remote host |
Hyper Text Transfer Protocol | HTTP | Used for the web to send documents that are encoded in HTML. |
Network File Services | NFS | Allows network nodes or workstations to access files and drives as if they were their own. |
Simple Mail Transfer Protocol | SMTP | Used to send Email over a network |
Telnet | Used to connect to a host and emulate a terminal that the remote server can recognize |
Introduction to TCP/IP Networks:
TCP/IP-based networks play an increasingly important role in computer networks. Perhaps one reason for their appeal is that they are based on an open specification that is not controlled by any vendor.
What Is TCP/IP?
TCP stands for Transmission Control Protocol and IP stands for Internet Protocol. The term TCP/IP is not limited just to these two protocols, however. Frequently, the term TCP/IP is used to refer to a group of protocols related to the TCP and IP protocols such as the User Datagram Protocol (UDP), File Transfer Protocol (FTP), Terminal Emulation Protocol (TELNET), and so on.
The Origins of TCP/IP
In the late 1960s, DARPA (the Defense Advanced Research Project Agency), in the United States, noticed that there was a rapid proliferation of computers in military communications. Computers, because they can be easily programmed, provide flexibility in achieving network functions that is not available with other types of communications equipment. The computers then used in military communications were manufactured by different vendors and were designed to interoperate with computers from that vendor only. Vendors used proprietary protocols in their communications equipment. The military had a multi vendor network but no common protocol to support the heterogeneous equipment from different vendors
Net work Cables and Stuff:
In the network you will commonly find three types of cables used these are the, coaxial cable, fiber optic and twisted pair.
Thick Coaxial Cable
This type cable is usually yellow in color and used in what is called thicknets, and has two conductors. This coax can be used in 500-meter lengths. The cable itself is made up of a solid center wire with a braided metal shield and plastic sheathing protecting the rest of the wire.
Thin Coaxial Cable
As with the thick coaxial cable is used in thicknets the thin version is used in thinnets. This type cable is also used called or referred to as RG-58. The cable is really just a cheaper version of the thick cable.
Fiber Optic Cable
As we all know fiber optics are pretty darn cool and not cheap. This cable is smaller and can carry a vast amount of information fast and over long distances.
Twisted Pair Cables
These come in two flavors of unshielded and shielded Shielded Twisted Pair (STP)
Is more common in high-speed networks. The biggest difference you will see in the UTP and STP is that the STP use's metallic shield wrapping to protect the wire from interference.
-Something else to note about these cables is that they are defined in numbers also. The bigger the number the better the protection from interference. Most networks should go with no less than a CAT 3 and CAT 5 is most recommended.
-Now you know about cables we need to know about connectors. This is pretty important and you will most likely need the RJ-45 connector. This is the cousin of the phone jack connector and looks real similar with the exception that the RJ-45 is bigger. Most commonly your connector are in two flavors and this is BNC (Bayonet Naur Connector) used in thicknets and the RJ-45 used in smaller networks using UTP/STP.
-Something else to note about these cables is that they are defined in numbers also. The bigger the number the better the protection from interference. Most networks should go with no less than a CAT 3 and CAT 5 is most recommended.
-Now you know about cables we need to know about connectors. This is pretty important and you will most likely need the RJ-45 connector. This is the cousin of the phone jack connector and looks real similar with the exception that the RJ-45 is bigger. Most commonly your connector are in two flavors and this is BNC (Bayonet Naur Connector) used in thicknets and the RJ-45 used in smaller networks using UTP/STP.
Unshielded Twisted Pair (UTP)
This is the most popular form of cables in the network and the cheapest form that you can go with. The UTP has four pairs of wires and all inside plastic sheathing. The biggest reason that we call it Twisted Pair is to protect the wires from interference from themselves. Each wire is only protected with a thin plastic sheath.
Ethernet Cabling
Now to familiarize you with more on the Ethernet and it's cabling we need to look at the 10's. 10Base2, is considered the thin Ethernet, thinnet, and thinwire which uses light coaxial cable to create a 10 Mbps network. The cable segments in this network can't be over 185 meters in length. These cables connect with the BNC connector. Also as a note these unused connection must have a terminator, which will be a 50-ohm terminator.
10Base5, this is considered a thicknet and is used with coaxial cable arrangement such as the BNC connector. The good side to the coaxial cable is the high-speed transfer and cable segments can be up to 500 meters between nodes/workstations. You will typically see the same speed as the 10Base2 but larger cable lengths for more versatility.
10BaseT, the “T” stands for twisted as in UTP (Unshielded Twisted Pair) and uses this for 10Mbps of transfer. The down side to this is you can only have cable lengths of 100 meters between nodes/workstations. The good side to this network is they are easy to set up and cheap! This is why they are so common an ideal for small offices or homes.
100BaseT, is considered Fast Ethernet uses STP (Shielded Twisted Pair) reaching data transfer of 100Mbps. This system is a little more expensive but still remains popular as the 10BaseT and cheaper than most other type networks. This on of course would be the cheap fast version.
10BaseF, this little guy has the advantage of fiber optics and the F stands for just that. This arrangement is a little more complicated and uses special connectors and NIC's along with hubs to create its network. Pretty darn neat and not to cheap on the wallet.
An important part of designing and installing an Ethernet is selecting the appropriate Ethernet medium. There are four major types of media in use today: Thickwire for 10BASE5 networks, thin coax for 10BASE2 networks, unshielded twisted pair (UTP) for 10BASE-T networks and fiber optic for 10BASE-FL or Fiber-Optic Inter-Repeater Link (FOIRL) networks. This wide variety of media reflects the evolution of Ethernet and also points to the technology's flexibility. Thickwire was one of the first cabling systems used in Ethernet but was expensive and difficult to use. This evolved to thin coax, which is easier to work with and less expensive.
10Base5, this is considered a thicknet and is used with coaxial cable arrangement such as the BNC connector. The good side to the coaxial cable is the high-speed transfer and cable segments can be up to 500 meters between nodes/workstations. You will typically see the same speed as the 10Base2 but larger cable lengths for more versatility.
10BaseT, the “T” stands for twisted as in UTP (Unshielded Twisted Pair) and uses this for 10Mbps of transfer. The down side to this is you can only have cable lengths of 100 meters between nodes/workstations. The good side to this network is they are easy to set up and cheap! This is why they are so common an ideal for small offices or homes.
100BaseT, is considered Fast Ethernet uses STP (Shielded Twisted Pair) reaching data transfer of 100Mbps. This system is a little more expensive but still remains popular as the 10BaseT and cheaper than most other type networks. This on of course would be the cheap fast version.
10BaseF, this little guy has the advantage of fiber optics and the F stands for just that. This arrangement is a little more complicated and uses special connectors and NIC's along with hubs to create its network. Pretty darn neat and not to cheap on the wallet.
An important part of designing and installing an Ethernet is selecting the appropriate Ethernet medium. There are four major types of media in use today: Thickwire for 10BASE5 networks, thin coax for 10BASE2 networks, unshielded twisted pair (UTP) for 10BASE-T networks and fiber optic for 10BASE-FL or Fiber-Optic Inter-Repeater Link (FOIRL) networks. This wide variety of media reflects the evolution of Ethernet and also points to the technology's flexibility. Thickwire was one of the first cabling systems used in Ethernet but was expensive and difficult to use. This evolved to thin coax, which is easier to work with and less expensive.
Network Topologies:
What is a Network topology?
A network topology is the geometric arrangement of nodes and cable links in a LAN,
There are three topology's to think about when you get into networks. These are the star, rind, and the bus.
Star, in a star topology each node has a dedicated set of wires connecting it to a central network hub. Since all traffic passes through the hub, the hub becomes a central point for isolating network problems and gathering network statistics.
Ring, a ring topology features a logically closed loop. Data packets travel in a single direction around the ring from one network device to the next. Each network device acts as a repeater, meaning it regenerates the signal
Bus, the bus topology, each node (computer, server, peripheral etc.) attaches directly to a common cable. This topology most often serves as the backbone for a network. In some instances, such as in classrooms or labs, a bus will connect small workgroups
There are three topology's to think about when you get into networks. These are the star, rind, and the bus.
Star, in a star topology each node has a dedicated set of wires connecting it to a central network hub. Since all traffic passes through the hub, the hub becomes a central point for isolating network problems and gathering network statistics.
Ring, a ring topology features a logically closed loop. Data packets travel in a single direction around the ring from one network device to the next. Each network device acts as a repeater, meaning it regenerates the signal
Bus, the bus topology, each node (computer, server, peripheral etc.) attaches directly to a common cable. This topology most often serves as the backbone for a network. In some instances, such as in classrooms or labs, a bus will connect small workgroups
Collisions:
Ethernet is a shared media, so there are rules for sending packets of data to avoid conflicts and protect data integrity. Nodes determine when the network is available for sending packets. It is possible that two nodes at different locations attempt to send data at the same time. When both PCs are transferring a packet to the network at the same time, a collision will result.
Minimizing collisions is a crucial element in the design and operation of networks. Increased collisions are often the result of too many users on the network, which results in a lot of contention for network bandwidth. This can slow the performance of the network from the user's point of view. Segmenting the network, where a network is divided into different pieces joined together logically with a bridge or switch, is one way of reducing an overcrowded network.
Minimizing collisions is a crucial element in the design and operation of networks. Increased collisions are often the result of too many users on the network, which results in a lot of contention for network bandwidth. This can slow the performance of the network from the user's point of view. Segmenting the network, where a network is divided into different pieces joined together logically with a bridge or switch, is one way of reducing an overcrowded network.
Ethernet Products:
The standards and technology that have just been discussed help define the specific products that network managers use to build Ethernet networks. The following text discusses the key products needed to build an Ethernet LAN.
Transceivers
Transceivers are used to connect nodes to the various Ethernet media. Most computers and network interface cards contain a built-in 10BASE-T or 10BASE2 transceiver, allowing them to be connected directly to Ethernet without requiring an external transceiver. Many Ethernet devices provide an AUI connector to allow the user to connect to any media type via an external transceiver. The AUI connector consists of a 15-pin D-shell type connector, female on the computer side, male on the transceiver side. Thickwire (10BASE5) cables also use transceivers to allow connections.
For Fast Ethernet networks, a new interface called the MII (Media Independent Interface) was developed to offer a flexible way to support 100 Mbps connections. The MII is a popular way to connect 100BASE-FX links to copper-based Fast Ethernet devices.
For Fast Ethernet networks, a new interface called the MII (Media Independent Interface) was developed to offer a flexible way to support 100 Mbps connections. The MII is a popular way to connect 100BASE-FX links to copper-based Fast Ethernet devices.
Network Interface Cards:
Network interface cards, commonly referred to as NICs, and are used to connect a PC to a network. The NIC provides a physical connection between the networking cable and the computer's internal bus. Different computers have different bus architectures; PCI bus master slots are most commonly found on 486/Pentium PCs and ISA expansion slots are commonly found on 386 and older PCs. NICs come in three basic varieties: 8-bit, 16-bit, and 32-bit. The larger the number of bits that can be transferred to the NIC, the faster the NIC can transfer data to the network cable.
Many NIC adapters comply with Plug-n-Play specifications. On these systems, NICs are automatically configured without user intervention, while on non-Plug-n-Play systems, configuration is done manually through a setup program and/or DIP switches.
Cards are available to support almost all networking standards, including the latest Fast Ethernet environment. Fast Ethernet NICs are often 10/100 capable, and will automatically set to the appropriate speed. Full duplex networking is another option, where a dedicated connection to a switch allows a NIC to operate at twice the speed.
Many NIC adapters comply with Plug-n-Play specifications. On these systems, NICs are automatically configured without user intervention, while on non-Plug-n-Play systems, configuration is done manually through a setup program and/or DIP switches.
Cards are available to support almost all networking standards, including the latest Fast Ethernet environment. Fast Ethernet NICs are often 10/100 capable, and will automatically set to the appropriate speed. Full duplex networking is another option, where a dedicated connection to a switch allows a NIC to operate at twice the speed.
Hubs/Repeaters:
Hubs/repeaters are used to connect together two or more Ethernet segments of any media type. In larger designs, signal quality begins to deteriorate as segments exceed their maximum length. Hubs provide the signal amplification required to allow a segment to be extended a greater distance. A hub takes any incoming signal and repeats it out all ports.
Ethernet hubs are necessary in star topologies such as 10BASE-T. A multi-port twisted pair hub allows several point-to-point segments to be joined into one network. One end of the point-to-point link is attached to the hub and the other is attached to the computer. If the hub is attached to a backbone, then all computers at the end of the twisted pair segments can communicate with all the hosts on the backbone. The number and type of hubs in any one-collision domain is limited by the Ethernet rules. These repeater rules are discussed in more detail later.
Ethernet hubs are necessary in star topologies such as 10BASE-T. A multi-port twisted pair hub allows several point-to-point segments to be joined into one network. One end of the point-to-point link is attached to the hub and the other is attached to the computer. If the hub is attached to a backbone, then all computers at the end of the twisted pair segments can communicate with all the hosts on the backbone. The number and type of hubs in any one-collision domain is limited by the Ethernet rules. These repeater rules are discussed in more detail later.
Network Type | Max Nodes Per Segment | Max Distance Per Segment |
10BASE-T 10BASE2 10BASE5 10BASE-FL | 2 30 100 2 | 100m 185m 500m 2000m |
Adding Speed:
While repeaters allow LANs to extend beyond normal distance limitations, they still limit the number of nodes that can be supported. Bridges and switches, however, allow LANs to grow significantly larger by virtue of their ability to support full Ethernet segments on each port. Additionally, bridges and switches selectively filter network traffic to only those packets needed on each segment - this significantly increases throughput on each segment and on the overall network. By providing better performance and more flexibility for network topologies, bridges and switches will continue to gain popularity among network managers.
Bridges:
The function of a bridge is to connect separate networks together. Bridges connect different networks types (such as Ethernet and Fast Ethernet) or networks of the same type. Bridges map the Ethernet addresses of the nodes residing on each network segment and allow only necessary traffic to pass through the bridge. When a packet is received by the bridge, the bridge determines the destination and source segments. If the segments are the same, the packet is dropped ("filtered"); if the segments are different, then the packet is "forwarded" to the correct segment. Additionally, bridges do not forward bad or misaligned packets.
Bridges are also called "store-and-forward" devices because they look at the whole Ethernet packet before making filtering or forwarding decisions. Filtering packets, and regenerating forwarded packets enable bridging technology to split a network into separate collision domains. This allows for greater distances and more repeaters to be used in the total network design.
Bridges are also called "store-and-forward" devices because they look at the whole Ethernet packet before making filtering or forwarding decisions. Filtering packets, and regenerating forwarded packets enable bridging technology to split a network into separate collision domains. This allows for greater distances and more repeaters to be used in the total network design.
Ethernet Switches:
Ethernet switches are an expansion of the concept in Ethernet bridging. LAN switches can link four, six, ten or more networks together, and have two basic architectures: cut-through and store-and-forward. In the past, cut-through switches were faster because they examined the packet destination address only before forwarding it on to its destination segment. A store-and-forward switch, on the other hand, accepts and analyzes the entire packet before forwarding it to its destination.
It takes more time to examine the entire packet, but it allows the switch to catch certain packet errors and keep them from propagating through the network. Both cut-through and store-and-forward switches separate a network into collision domains, allowing network design rules to be extended. Each of the segments attached to an Ethernet switch has a full 10 Mbps of bandwidth shared by fewer users, which results in better performance (as opposed to hubs that only allow bandwidth sharing from a single Ethernet). Newer switches today offer high-speed links, FDDI, Fast Ethernet or ATM. These are used to link switches together or give added bandwidth to high-traffic servers. A network composed of a number of switches linked together via uplinks is termed a "collapsed backbone" network.
It takes more time to examine the entire packet, but it allows the switch to catch certain packet errors and keep them from propagating through the network. Both cut-through and store-and-forward switches separate a network into collision domains, allowing network design rules to be extended. Each of the segments attached to an Ethernet switch has a full 10 Mbps of bandwidth shared by fewer users, which results in better performance (as opposed to hubs that only allow bandwidth sharing from a single Ethernet). Newer switches today offer high-speed links, FDDI, Fast Ethernet or ATM. These are used to link switches together or give added bandwidth to high-traffic servers. A network composed of a number of switches linked together via uplinks is termed a "collapsed backbone" network.
Routers:
Routers filter out network traffic by specific protocol rather than by packet address. Routers also divide networks logically instead of physically. An IP router can divide a network into various subnets so that only traffic destined for particular IP addresses can pass between segments. Network speed often decreases due to this type of intelligent forwarding. Such filtering takes more time than that exercised in a switch or bridge, which only looks at the Ethernet address. However, in more complex networks, overall efficiency is improved by using routers.
What is a network firewall?
A firewall is a system or group of systems that enforces an access control policy between two networks. The actual means by which this is accomplished varies widely, but in principle, the firewall can be thought of as a pair of mechanisms: one which exists to block traffic, and the other which exists to permit traffic. Some firewalls place a greater emphasis on blocking traffic, while others emphasize permitting traffic. Probably the most important thing to recognize about a firewall is that it implements an access control policy. If you don't have a good idea of what kind of access you want to allow or to deny, a firewall really won't help you. It's also important to recognize that the firewall's configuration, because it is a mechanism for enforcing policy, imposes its policy on everything behind it. Administrators for firewalls managing the connectivity for a large number of hosts therefore have a heavy responsibility.
Network Design Criteria:
Ethernets and Fast Ethernets have design rules that must be followed in order to function correctly. Maximum number of nodes, number of repeaters and maximum segment distances are defined by the electrical and mechanical design properties of each type of Ethernet and Fast Ethernet media.
A network using repeaters, for instance, functions with the timing constraints of Ethernet. Although electrical signals on the Ethernet media travel near the speed of light, it still takes a finite time for the signal to travel from one end of a large Ethernet to another. The Ethernet standard assumes it will take roughly 50 microseconds for a signal to reach its destination.
Ethernet is subject to the "5-4-3" rule of repeater placement: the network can only have five segments connected; it can only use four repeaters; and of the five segments, only three can have users attached to them; the other two must be inter-repeater links.
If the design of the network violates these repeater and placement rules, then timing guidelines will not be met and the sending station will resend that packet. This can lead to lost packets and excessive resent packets, which can slow network performance and create trouble for applications. Fast Ethernet has modified repeater rules, since the minimum packet size takes less time to transmit than regular Ethernet. The length of the network links allows for a fewer number of repeaters. In Fast Ethernet networks, there are two classes of repeaters. Class I repeaters have a latency of 0.7 microseconds or less and are limited to one repeater per network. Class II repeaters have a latency of 0.46 microseconds or less and are limited to two repeaters per network. The following are the distance (diameter) characteristics for these types of Fast Ethernet repeater combinations:
A network using repeaters, for instance, functions with the timing constraints of Ethernet. Although electrical signals on the Ethernet media travel near the speed of light, it still takes a finite time for the signal to travel from one end of a large Ethernet to another. The Ethernet standard assumes it will take roughly 50 microseconds for a signal to reach its destination.
Ethernet is subject to the "5-4-3" rule of repeater placement: the network can only have five segments connected; it can only use four repeaters; and of the five segments, only three can have users attached to them; the other two must be inter-repeater links.
If the design of the network violates these repeater and placement rules, then timing guidelines will not be met and the sending station will resend that packet. This can lead to lost packets and excessive resent packets, which can slow network performance and create trouble for applications. Fast Ethernet has modified repeater rules, since the minimum packet size takes less time to transmit than regular Ethernet. The length of the network links allows for a fewer number of repeaters. In Fast Ethernet networks, there are two classes of repeaters. Class I repeaters have a latency of 0.7 microseconds or less and are limited to one repeater per network. Class II repeaters have a latency of 0.46 microseconds or less and are limited to two repeaters per network. The following are the distance (diameter) characteristics for these types of Fast Ethernet repeater combinations:
Fast Ethernet | Copper | Fiber |
No Repeaters One Class I Repeater One Class II Repeater Two Class II Repeaters | 100m 200m 200m 205m | 412m* 272m 272m 228m |
* Full Duplex Mode 2 km
When conditions require greater distances or an increase in the number of nodes/repeaters, then a bridge, router or switch can be used to connect multiple networks together. These devices join two or more separate networks, allowing network design criteria to be restored. Switches allow network designers to build large networks that function well. The reduction in costs of bridges and switches reduces the impact of repeater rules on network design.
Each network connected via one of these devices is referred to as a separate collision domain in the overall network.
Types of Servers:
Device Servers
A device server is defined as a specialized, network-based hardware device designed to perform a single or specialized set of server functions. It is characterized by a minimal operating architecture that requires no per seat network operating system license, and client access that is independent of any operating system or proprietary protocol. In addition the device server is a "closed box," delivering extreme ease of installation, minimal maintenance, and can be managed by the client remotely via a Web browser.
Print servers, terminal servers, remote access servers and network time servers are examples of device servers which are specialized for particular functions. Each of these types of servers has unique configuration attributes in hardware or software that help them to perform best in their particular arena.
Print servers, terminal servers, remote access servers and network time servers are examples of device servers which are specialized for particular functions. Each of these types of servers has unique configuration attributes in hardware or software that help them to perform best in their particular arena.
Print Servers
Print servers allow printers to be shared by other users on the network. Supporting either parallel and/or serial interfaces, a print server accepts print jobs from any person on the network using supported protocols and manages those jobs on each appropriate printer.
Print servers generally do not contain a large amount of memory; printers simply store information in a queue. When the desired printer becomes available, they allow the host to transmit the data to the appropriate printer port on the server. The print server can then simply queue and print each job in the order in which print requests are received, regardless of protocol used or the size of the job.
Print servers generally do not contain a large amount of memory; printers simply store information in a queue. When the desired printer becomes available, they allow the host to transmit the data to the appropriate printer port on the server. The print server can then simply queue and print each job in the order in which print requests are received, regardless of protocol used or the size of the job.
Multiport Device Servers
Devices that are attached to a network through a multiport device server can be shared between terminals and hosts at both the local site and throughout the network. A single terminal may be connected to several hosts at the same time (in multiple concurrent sessions), and can switch between them. Multiport device servers are also used to network devices that have only serial outputs. A connection between serial ports on different servers is opened, allowing data to move between the two devices.
Given its natural translation ability, a multi-protocol multiport device server can perform conversions between the protocols it knows, like LAT and TCP/IP. While server bandwidth is not adequate for large file transfers, it can easily handle host-to-host inquiry/response applications, electronic mailbox checking, etc. And it is far more economical than the alternatives of acquiring expensive host software and special-purpose converters. Multiport device and print servers give their users greater flexibility in configuring and managing their networks.
Whether it is moving printers and other peripherals from one network to another, expanding the dimensions of interoperability or preparing for growth, multiport device servers can fulfill your needs, all without major rewiring.
Given its natural translation ability, a multi-protocol multiport device server can perform conversions between the protocols it knows, like LAT and TCP/IP. While server bandwidth is not adequate for large file transfers, it can easily handle host-to-host inquiry/response applications, electronic mailbox checking, etc. And it is far more economical than the alternatives of acquiring expensive host software and special-purpose converters. Multiport device and print servers give their users greater flexibility in configuring and managing their networks.
Whether it is moving printers and other peripherals from one network to another, expanding the dimensions of interoperability or preparing for growth, multiport device servers can fulfill your needs, all without major rewiring.
Access Servers
While Ethernet is limited to a geographic area, remote users such as traveling sales people need access to network-based resources. Remote LAN access, or remote access, is a popular way to provide this connectivity. Access servers use telephone services to link a user or office with an office network. Dial-up remote access solutions such as ISDN or asynchronous dial introduce more flexibility. Dial-up remote access offers both the remote office and the remote user the economy and flexibility of "pay as you go" telephone services. ISDN is a special telephone service that offers three channels, two 64 Kbps "B" channels for user data and a "D" channel for setting up the connection. With ISDN, the B channels can be combined for double bandwidth or separated for different applications or users. With asynchronous remote access, regular telephone lines are combined with modems and remote access servers to allow users and networks to dial anywhere in the world and have data access. Remote access servers provide connection points for both dial-in and dial-out applications on the network to which they are attached. These hybrid devices route and filter protocols and offer other services such as modem pooling and terminal/printer services. For the remote PC user, one can connect from any available telephone jack (RJ45), including those in a hotel rooms or on most airplanes.
Network Time Servers
A network time server is a server specialized in the handling of timing information from sources such as satellites or radio broadcasts and is capable of providing this timing data to its attached network. Specialized protocols such as NTP or udp/time allow a time server to communicate to other network nodes ensuring that activities that must be coordinated according to their time of execution are synchronized correctly. GPS satellites are one source of information that can allow global installations to achieve constant timing.
IP Addressing:
An IP (Internet Protocol) address is a unique identifier for a node or host connection on an IP network. An IP address is a 32 bit binary number usually represented as 4 decimal values, each representing 8 bits, in the range 0 to 255 (known as octets) separated by decimal points. This is known as "dotted decimal" notation.
Example: 140.179.220.200
It is sometimes useful to view the values in their binary form.
140 .179 .220 .200
10001100.10110011.11011100.11001000
Every IP address consists of two parts, one identifying the network and one identifying the node. The Class of the address and the subnet mask determine which part belongs to the network address and which part belongs to the node address.
Example: 140.179.220.200
It is sometimes useful to view the values in their binary form.
140 .179 .220 .200
10001100.10110011.11011100.11001000
Every IP address consists of two parts, one identifying the network and one identifying the node. The Class of the address and the subnet mask determine which part belongs to the network address and which part belongs to the node address.
Address Classes:
There are 5 different address classes. You can determine which class any IP address is in by examining the first 4 bits of the IP address.
Class A addresses begin with 0xxx, or 1 to 126 decimal.
Class B addresses begin with 10xx, or 128 to 191 decimal.
Class C addresses begin with 110x, or 192 to 223 decimal.
Class D addresses begin with 1110, or 224 to 239 decimal.
Class E addresses begin with 1111, or 240 to 254 decimal.
Addresses beginning with 01111111, or 127 decimal, are reserved for loopback and for internal testing on a local machine. [You can test this: you should always be able to ping 127.0.0.1, which points to yourself] Class D addresses are reserved for multicasting. Class E addresses are reserved for future use. They should not be used for host addresses.
Now we can see how the Class determines, by default, which part of the IP address belongs to the network (N) and which part belongs to the node (n).
Class A -- NNNNNNNN.nnnnnnnn.nnnnnnn.nnnnnnn
Class B -- NNNNNNNN.NNNNNNNN.nnnnnnnn.nnnnnnnn
Class C -- NNNNNNNN.NNNNNNNN.NNNNNNNN.nnnnnnnn
In the example, 140.179.220.200 is a Class B address so by default the Network part of the address (also known as the Network Address) is defined by the first two octets (140.179.x.x) and the node part is defined by the last 2 octets (x.x.220.200).
In order to specify the network address for a given IP address, the node section is set to all "0"s. In our example, 140.179.0.0 specifies the network address for 140.179.220.200. When the node section is set to all "1"s, it specifies a broadcast that is sent to all hosts on the network. 140.179.255.255 specifies the example broadcast address. Note that this is true regardless of the length of the node section.
Class A addresses begin with 0xxx, or 1 to 126 decimal.
Class B addresses begin with 10xx, or 128 to 191 decimal.
Class C addresses begin with 110x, or 192 to 223 decimal.
Class D addresses begin with 1110, or 224 to 239 decimal.
Class E addresses begin with 1111, or 240 to 254 decimal.
Addresses beginning with 01111111, or 127 decimal, are reserved for loopback and for internal testing on a local machine. [You can test this: you should always be able to ping 127.0.0.1, which points to yourself] Class D addresses are reserved for multicasting. Class E addresses are reserved for future use. They should not be used for host addresses.
Now we can see how the Class determines, by default, which part of the IP address belongs to the network (N) and which part belongs to the node (n).
Class A -- NNNNNNNN.nnnnnnnn.nnnnnnn.nnnnnnn
Class B -- NNNNNNNN.NNNNNNNN.nnnnnnnn.nnnnnnnn
Class C -- NNNNNNNN.NNNNNNNN.NNNNNNNN.nnnnnnnn
In the example, 140.179.220.200 is a Class B address so by default the Network part of the address (also known as the Network Address) is defined by the first two octets (140.179.x.x) and the node part is defined by the last 2 octets (x.x.220.200).
In order to specify the network address for a given IP address, the node section is set to all "0"s. In our example, 140.179.0.0 specifies the network address for 140.179.220.200. When the node section is set to all "1"s, it specifies a broadcast that is sent to all hosts on the network. 140.179.255.255 specifies the example broadcast address. Note that this is true regardless of the length of the node section.
Private Subnets:
There are three IP network addresses reserved for private networks. The addresses are 10.0.0.0/8, 172.16.0.0/12, and 192.168.0.0/16. They can be used by anyone setting up internal IP networks, such as a lab or home LAN behind a NAT or proxy server or a router. It is always safe to use these because routers on the Internet will never forward packets coming from these addresses
Subnetting an IP Network can be done for a variety of reasons, including organization, use of different physical media (such as Ethernet, FDDI, WAN, etc.), preservation of address space, and security. The most common reason is to control network traffic. In an Ethernet network, all nodes on a segment see all the packets transmitted by all the other nodes on that segment. Performance can be adversely affected under heavy traffic loads, due to collisions and the resulting retransmissions. A router is used to connect IP networks to minimize the amount of traffic each segment must receive.
Subnetting an IP Network can be done for a variety of reasons, including organization, use of different physical media (such as Ethernet, FDDI, WAN, etc.), preservation of address space, and security. The most common reason is to control network traffic. In an Ethernet network, all nodes on a segment see all the packets transmitted by all the other nodes on that segment. Performance can be adversely affected under heavy traffic loads, due to collisions and the resulting retransmissions. A router is used to connect IP networks to minimize the amount of traffic each segment must receive.
Subnet Masking
Applying a subnet mask to an IP address allows you to identify the network and node parts of the address. The network bits are represented by the 1s in the mask, and the node bits are represented by the 0s. Performing a bitwise logical AND operation between the IP address and the subnet mask results in the Network Address or Number.
For example, using our test IP address and the default Class B subnet mask, we get:
10001100.10110011.11110000.11001000 140.179.240.200 Class B IP Address
11111111.11111111.00000000.00000000 255.255.000.000 Default Class B Subnet Mask
10001100.10110011.00000000.00000000 140.179.000.000 Network Address
For example, using our test IP address and the default Class B subnet mask, we get:
10001100.10110011.11110000.11001000 140.179.240.200 Class B IP Address
11111111.11111111.00000000.00000000 255.255.000.000 Default Class B Subnet Mask
10001100.10110011.00000000.00000000 140.179.000.000 Network Address
Default subnet masks:
Class A - 255.0.0.0 - 11111111.00000000.00000000.00000000
Class B - 255.255.0.0 - 11111111.11111111.00000000.00000000
Class C - 255.255.255.0 - 11111111.11111111.11111111.00000000
CIDR -- Classless InterDomain Routing.
CIDR was invented several years ago to keep the internet from running out of IP addresses. The "classful" system of allocating IP addresses can be very wasteful; anyone who could reasonably show a need for more that 254 host addresses was given a Class B address block of 65533 host addresses. Even more wasteful were companies and organizations that were allocated Class A address blocks, which contain over 16 Million host addresses! Only a tiny percentage of the allocated Class A and Class B address space has ever been actually assigned to a host computer on the Internet.
People realized that addresses could be conserved if the class system was eliminated. By accurately allocating only the amount of address space that was actually needed, the address space crisis could be avoided for many years. This was first proposed in 1992 as a scheme called Supernetting.
The use of a CIDR notated address is the same as for a Classful address. Classful addresses can easily be written in CIDR notation (Class A = /8, Class B = /16, and Class C = /24)
It is currently almost impossible for an individual or company to be allocated their own IP address blocks. You will simply be told to get them from your ISP. The reason for this is the ever-growing size of the internet routing table. Just 5 years ago, there were less than 5000 network routes in the entire Internet. Today, there are over 90,000. Using CIDR, the biggest ISPs are allocated large chunks of address space (usually with a subnet mask of /19 or even smaller); the ISP's customers (often other, smaller ISPs) are then allocated networks from the big ISP's pool. That way, all the big ISP's customers (and their customers, and so on) are accessible via 1 network route on the Internet.
It is expected that CIDR will keep the Internet happily in IP addresses for the next few years at least. After that, IPv6, with 128 bit addresses, will be needed. Under IPv6, even sloppy address allocation would comfortably allow a billion unique IP addresses for every person on earth
Class B - 255.255.0.0 - 11111111.11111111.00000000.00000000
Class C - 255.255.255.0 - 11111111.11111111.11111111.00000000
CIDR -- Classless InterDomain Routing.
CIDR was invented several years ago to keep the internet from running out of IP addresses. The "classful" system of allocating IP addresses can be very wasteful; anyone who could reasonably show a need for more that 254 host addresses was given a Class B address block of 65533 host addresses. Even more wasteful were companies and organizations that were allocated Class A address blocks, which contain over 16 Million host addresses! Only a tiny percentage of the allocated Class A and Class B address space has ever been actually assigned to a host computer on the Internet.
People realized that addresses could be conserved if the class system was eliminated. By accurately allocating only the amount of address space that was actually needed, the address space crisis could be avoided for many years. This was first proposed in 1992 as a scheme called Supernetting.
The use of a CIDR notated address is the same as for a Classful address. Classful addresses can easily be written in CIDR notation (Class A = /8, Class B = /16, and Class C = /24)
It is currently almost impossible for an individual or company to be allocated their own IP address blocks. You will simply be told to get them from your ISP. The reason for this is the ever-growing size of the internet routing table. Just 5 years ago, there were less than 5000 network routes in the entire Internet. Today, there are over 90,000. Using CIDR, the biggest ISPs are allocated large chunks of address space (usually with a subnet mask of /19 or even smaller); the ISP's customers (often other, smaller ISPs) are then allocated networks from the big ISP's pool. That way, all the big ISP's customers (and their customers, and so on) are accessible via 1 network route on the Internet.
It is expected that CIDR will keep the Internet happily in IP addresses for the next few years at least. After that, IPv6, with 128 bit addresses, will be needed. Under IPv6, even sloppy address allocation would comfortably allow a billion unique IP addresses for every person on earth
Examining your network with commands:
Ping
PING is used to check for a response from another computer on the network. It can tell you a great deal of information about the status of the network and the computers you are communicating with.
Ping returns different responses depending on the computer in question. The responses are similar depending on the options used.
Ping uses IP to request a response from the host. It does not use TCP
.It takes its name from a submarine sonar search - you send a short sound burst and listen for an echo - a ping - coming back.
In an IP network, `ping' sends a short data burst - a single packet - and listens for a single packet in reply. Since this tests the most basic function of an IP network (delivery of single packet), it's easy to see how you can learn a lot from some `pings'.
To stop ping, type control-c. This terminates the program and prints out a nice summary of the number of packets transmitted, the number received, and the percentage of packets lost, plus the minimum, average, and maximum round-trip times of the packets.
Sample ping session
PING localhost (127.0.0.1): 56 data bytes
64 bytes from 127.0.0.1: icmp_seq=0 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=1 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=2 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=3 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=4 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=5 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=6 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=7 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=8 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=9 ttl=255 time=2 ms
localhost ping statistics
10 packets transmitted, 10 packets received, 0% packet loss
round-trip min/avg/max = 2/2/2 ms
meikro$
The Time To Live (TTL) field can be interesting. The main purpose of this is so that a packet doesn't live forever on the network and will eventually die when it is deemed "lost." But for us, it provides additional information. We can use the TTL to determine approximately how many router hops the packet has gone through. In this case it's 255 minus N hops, where N is the TTL of the returning Echo Replies. If the TTL field varies in successive pings, it could indicate that the successive reply packets are going via different routes, which isn't a great thing.
The time field is an indication of the round-trip time to get a packet to the remote host. The reply is measured in milliseconds. In general, it's best if round-trip times are under 200 milliseconds. The time it takes a packet to reach its destination is called latency. If you see a large variance in the round-trip times (which is called "jitter"), you are going to see poor performance talking to the host
PING is used to check for a response from another computer on the network. It can tell you a great deal of information about the status of the network and the computers you are communicating with.
Ping returns different responses depending on the computer in question. The responses are similar depending on the options used.
Ping uses IP to request a response from the host. It does not use TCP
.It takes its name from a submarine sonar search - you send a short sound burst and listen for an echo - a ping - coming back.
In an IP network, `ping' sends a short data burst - a single packet - and listens for a single packet in reply. Since this tests the most basic function of an IP network (delivery of single packet), it's easy to see how you can learn a lot from some `pings'.
To stop ping, type control-c. This terminates the program and prints out a nice summary of the number of packets transmitted, the number received, and the percentage of packets lost, plus the minimum, average, and maximum round-trip times of the packets.
Sample ping session
PING localhost (127.0.0.1): 56 data bytes
64 bytes from 127.0.0.1: icmp_seq=0 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=1 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=2 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=3 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=4 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=5 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=6 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=7 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=8 ttl=255 time=2 ms
64 bytes from 127.0.0.1: icmp_seq=9 ttl=255 time=2 ms
localhost ping statistics
10 packets transmitted, 10 packets received, 0% packet loss
round-trip min/avg/max = 2/2/2 ms
meikro$
The Time To Live (TTL) field can be interesting. The main purpose of this is so that a packet doesn't live forever on the network and will eventually die when it is deemed "lost." But for us, it provides additional information. We can use the TTL to determine approximately how many router hops the packet has gone through. In this case it's 255 minus N hops, where N is the TTL of the returning Echo Replies. If the TTL field varies in successive pings, it could indicate that the successive reply packets are going via different routes, which isn't a great thing.
The time field is an indication of the round-trip time to get a packet to the remote host. The reply is measured in milliseconds. In general, it's best if round-trip times are under 200 milliseconds. The time it takes a packet to reach its destination is called latency. If you see a large variance in the round-trip times (which is called "jitter"), you are going to see poor performance talking to the host
NSLOOKUP
NSLOOKUP is an application that facilitates looking up hostnames on the network. It can reveal the IP address of a host or, using the IP address, return the host name.
It is very important when troubleshooting problems on a network that you can verify the components of the networking process. Nslookup allows this by revealing details within the infrastructure.
It is very important when troubleshooting problems on a network that you can verify the components of the networking process. Nslookup allows this by revealing details within the infrastructure.
NETSTAT
NETSTAT is used to look up the various active connections within a computer. It is helpful to understand what computers or networks you are connected to. This allows you to further investigate problems. One host may be responding well but another may be less responsive.
IPconfig
This is a Microsoft windows NT, 2000 command. It is very useful in determining what could be wrong with a network.
This command when used with the /all switch, reveal enormous amounts of troubleshooting information within the system.
Windows 2000 IP Configuration
Host Name . . . . . . . . . . . . : cowder
Primary DNS Suffix . . . . . . . :
Node Type . . . . . . . . . . . . : Broadcast
IP Routing Enabled. . . . . . . . : No
WINS Proxy Enabled. . . . . . . . : No
WINS Proxy Enabled. . . . . . . . : No
Connection-specific DNS Suffix . :
Description . . . . . . . . . . . :
WAN (PPP/SLIP) Interface
Physical Address. . . . . . . . . : 00-53-45-00-00-00 begin_of_the_skype_highlighting 00-53-45-00-00-00 end_of_the_skype_highlighting
DHCP Enabled. . . . . . . . . . . : No
IP Address. . . . . . . . . . . . : 12.90.108.123
Subnet Mask . . . . . . . . . . . : 255.255.255.255
Default Gateway . . . . . . . . . : 12.90.108.125
DNS Servers . . . . . . . . . . . : 12.102.244.2
204.127.129.2
This command when used with the /all switch, reveal enormous amounts of troubleshooting information within the system.
Windows 2000 IP Configuration
Host Name . . . . . . . . . . . . : cowder
Primary DNS Suffix . . . . . . . :
Node Type . . . . . . . . . . . . : Broadcast
IP Routing Enabled. . . . . . . . : No
WINS Proxy Enabled. . . . . . . . : No
WINS Proxy Enabled. . . . . . . . : No
Connection-specific DNS Suffix . :
Description . . . . . . . . . . . :
WAN (PPP/SLIP) Interface
Physical Address. . . . . . . . . : 00-53-45-00-00-00 begin_of_the_skype_highlighting 00-53-45-00-00-00 end_of_the_skype_highlighting
DHCP Enabled. . . . . . . . . . . : No
IP Address. . . . . . . . . . . . : 12.90.108.123
Subnet Mask . . . . . . . . . . . : 255.255.255.255
Default Gateway . . . . . . . . . : 12.90.108.125
DNS Servers . . . . . . . . . . . : 12.102.244.2
204.127.129.2
Traceroute
Traceroute on Unix and Linux (or tracert in the Microsoft world) attempts to trace the current network path to a destination. Here is an example of a traceroute run to www.berkeley.edu:
$ traceroute www.berkeley.edu
traceroute to amber.Berkeley.EDU (128.32.25.12), 30 hops max, 40 byte packets
1 sf1-e3.wired.net (206.221.193.1) 3.135 ms 3.021 ms 3.616 ms
2 sf0-e2s2.wired.net (205.227.206.33) 1.829 ms 3.886 ms 2.772 ms
3 paloalto-cr10.bbnplanet.net (131.119.26.105) 5.327 ms 4.597 ms 5.729 ms
4 paloalto-br1.bbnplanet.net (131.119.0.193) 4.842 ms 4.615 ms 3.425 ms
5 sl-sj-2.sprintlink.net (4.0.1.66) 7.488 ms 38.804 ms 7.708 ms
6 144.232.8.81 (144.232.8.81) 6.560 ms 6.631 ms 6.565 ms
7 144.232.4.97 (144.232.4.97) 7.638 ms 7.948 ms 8.129 ms
8 144.228.146.50 (144.228.146.50) 9.504 ms 12.684 ms 16.648 ms
9 f5-0.inr-666-eva.berkeley.edu (198.128.16.21) 9.762 ms 10.611 ms 10.403 ms
10 f0-0.inr-107-eva.Berkeley.EDU (128.32.2.1) 11.478 ms 10.868 ms 9.367 ms
11 f8-0.inr-100-eva.Berkeley.EDU (128.32.235.100) 10.738 ms 11.693 ms 12.520 ms
$ traceroute www.berkeley.edu
traceroute to amber.Berkeley.EDU (128.32.25.12), 30 hops max, 40 byte packets
1 sf1-e3.wired.net (206.221.193.1) 3.135 ms 3.021 ms 3.616 ms
2 sf0-e2s2.wired.net (205.227.206.33) 1.829 ms 3.886 ms 2.772 ms
3 paloalto-cr10.bbnplanet.net (131.119.26.105) 5.327 ms 4.597 ms 5.729 ms
4 paloalto-br1.bbnplanet.net (131.119.0.193) 4.842 ms 4.615 ms 3.425 ms
5 sl-sj-2.sprintlink.net (4.0.1.66) 7.488 ms 38.804 ms 7.708 ms
6 144.232.8.81 (144.232.8.81) 6.560 ms 6.631 ms 6.565 ms
7 144.232.4.97 (144.232.4.97) 7.638 ms 7.948 ms 8.129 ms
8 144.228.146.50 (144.228.146.50) 9.504 ms 12.684 ms 16.648 ms
9 f5-0.inr-666-eva.berkeley.edu (198.128.16.21) 9.762 ms 10.611 ms 10.403 ms
10 f0-0.inr-107-eva.Berkeley.EDU (128.32.2.1) 11.478 ms 10.868 ms 9.367 ms
11 f8-0.inr-100-eva.Berkeley.EDU (128.32.235.100) 10.738 ms 11.693 ms 12.520 ms
