Frame Relay Layers

Frame Relay Layers

Frame Relay has only physical and data link layers.


Physical Layer

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

Data Link Layer

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

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

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

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

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

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

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



Extended Address

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


FRADs

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

VOFR

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

LMI

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


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

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

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

Virtual-Circuit Networks

Virtual-Circuit Networks.' Frame Relay and ATM


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

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

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


FRAME RELAY

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

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

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

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

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


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

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

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

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

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

3. Frame Relay allows bursty data.

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

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

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

Architecture

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

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

Permanent Versus Switched Virtual Circuits

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

Switches

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

SONET NETWORKS

SONET NETWORKS

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

Linear Networks

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

Multipoint Network

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

Automatic Protection Switching

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

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


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

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

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


Ring Networks

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

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

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

Bidirectional Line Switching Ring

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

Combination of Rings

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

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


VIRTUAL TRIBUTARIES

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

Types of VTs

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

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


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

Encapsulation

Encapsulation

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

Offsetting

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

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


STS MULTIPLEXING

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

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

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

Byte Interleaving

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

Concatenated Signal

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

Add/Drop Multiplexer

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

STS-l frame: line overhead

STS-l frame: line overhead

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

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

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

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

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

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

Synchronous Payload Envelope

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

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

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

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

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

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

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

SONET LAYERS

SONET LAYERS

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

Path Layer

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


Line Layer

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

Section Layer

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

Photonic Layer

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

Device-Layer Relationships

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

SONET FRAMES

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

Frame, Byte, and Bit Transmission

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


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

STS-1 Frame Format

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

Section Overhead

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

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

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

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

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

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

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

SONET/SDH

SONET/SDH

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

ARCHITECTURE

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


SONET Devices

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

STS Multtiplexer/Detnultiplexer

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

Regenerator

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


Add/drop Multiplexer

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

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

Terminals

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

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

Sections

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

Lines

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

Paths

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

GSM

GSM

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

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

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


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

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

IS-95

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

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

19.2 ksps x 64 cps = 1.228 Mcps

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

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

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

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

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


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

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

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

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

Wireless WAN

Wireless WANs: Cellular
and Satellite Networks
Telephone

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

CELLULAR TELEPHONY

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

Frequency-Reuse Principle

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

Transmitting

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

Receiving

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

Handoff

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

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

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

Roaming

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

First Generation

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

AMPS

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

Second Generation

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

D-AMPS

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

BLUETOOTH

BLUETOOTH

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

Blaatand translates to Bluetooth in English.

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

Architecture

Bluetooth defines two types of networks: piconet and scatternet.

Piconets

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

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

Scatternet

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

Bluetooth Devices

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

Radio Layer

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

Band

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

FHSS

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

Network Allocation Vector

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

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

Point Coordination Function (PCF)

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


Fragnentation

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

Wireless, Wireless LANs, LAN Arcitecture,IEEE 802.11

What is Wireless LAN, Wireless LAN overview.

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

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

Source : wikipedia

Wireless LANs

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

IEEE 802.11

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

Architecture

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

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

Extended Service Set

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

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

MAC Sublayer

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

Benifits of Wireless LANs -

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

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

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

CDMA

Code-Division Multiple Access (CDMA)

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

Analogy

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

Idea

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

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

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

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

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

Chips

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

CHANNELIZATION

CHANNELIZATION

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

Frequency-Division Multiple Access (FDMA)

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

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

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

Time-Division Multiple Access (TDMA)

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

Token Passing

Token Passing

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

Logical Ring

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

CSMA/CA

CSMA/CA and Wireless Networks

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

CONTROLLED ACCESS

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

Reservation

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

Polling

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

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

Select

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

Poll

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

Collision Avoidance

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

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

Interframe Space (IFS)

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

Contention Window

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

Acknowledgment

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

Procedure

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

Multiple Access

Multiple Access

In previous post we discussed data link control, a mechanism which provides a link with reliable communication. In the protocols we described, we assumed that there is an available dedicated link (or channel) between the sender and the receiver. This assumption may or may not be true. If, indeed, we have a dedicated link, as when we connect to the Internet using PPP as the data link control protocol, then the assumption is true and we do not need anything else. On the other hand, if we use our cellular phone to connect to another cellular phone, the channel (the band allocated to the vendor company) is not dedicated. A person a few feet away from us may be using the same channel to talk to her friend. We can consider the data link layer as two sublayers. The upper sublayer is responsible for data link control, and the lower sublayer is responsible for resolving access to the shared media. If the channel is dedicated, we do not need the lower sublayer. the IEEE has actually made this division for LANs. The upper sublayer that is responsible for flow and error control is called the logical link control (LLC) layer; the lower sublayer that is mostly responsible for multipleaccess resolution is called the media access control (MAC) layer. When nodes or stations are connected and use a common link, called a multipoint or broadcast link, we need a multiple-access protocol to coordinate access to the link. The problem of controlling the access to the medium is similar to the rules of speaking in an assembly. The procedures guarantee that the right to speak is upheld and ensure that two people do not speak at the same time, do not interrupt each other, do not monopolize the discussion, and so on. The situation is similar for multipoint networks. Many formal protocols have been devised to handle access to a shared link. We categorize them into three groups.

RANDOM ACCESS

In random access or contention methods, no station is superior to another station and none is assigned the control over another. No station permits, or does not permit,another station to send. At each instance, a station that has data to send uses a procedure defined by the protocol to make a decision on whether or not to send. This decision depends on the state of the medium (idle or busy). In other words, each station can transmit when it desires on the condition that it follows the predefined procedure, including the testing of the state of the medium. Two features give this method its name. First, there is no scheduled time for a station to transmit. Transmission is random among the stations. That is why these methods are called random access. Second, no rules specify which station should send next. Stations compete with one another to access the medium. That is why these methods are also called contention methods. In a random access method, each station has the right to the medium without being controlled by any other station. However, if more than one station tries to send, there is an access conflict--collision--and the frames will be either destroyed or modified. To avoid access conflict or to resolve it when it happens, each station follows a procedure that answers the following questions:

#When can the station access the medium?
#What can the station do if the medium is busy?
#How can the station determine the success or failure of the transmission?
#What can the station do if there is an access conflict?

The random access methods we study in this chapter have evolved from a very interesting protocol known as ALOHA, which used a very simple procedure called multiple access (MA). The method was improved with the addition of a procedure that forces the station to sense the medium before transmitting. This was called carrier sense multiple access. This method later evolved into two parallel methods: carrier sense multiple access with collision detection (CSMA/CD) and carrier sense multiple access with collision avoidance (CSMA/CA). CSMA/CD tells the station what to do when a collision is detected. CSMA/CA tries to avoid the collision.

Data Link Control

The two main functions of the data link layer are data link control and media access control. The first, data link control, deals with the design and procedures for communication between two adjacent nodes: node-to-node communication. We discuss this functionality in this chapter. The second function of the data link layer is media access control, or how to share the link.

Data link control functions include framing, flow and error control, and software- implemented protocols that provide smooth and reliable transmission of frames between nodes. In this chapter, we first discuss framing, or how to organize the bits that are carried by the physical layer. We then discuss flow and error control. A subset of this topic, techniques for error detection and correction. To implement data link control, we need protocols. Each protocol is a set of rules that need to be implemented in software and run by the two nodes involved in data

exchange at the data link layer. We discuss five protocols: two for noiseless (ideal) channels and three for noisy (real) channels. Those in the first category are not actually implemented, but provide a foundation for understanding the protocols in the second category. After discussing the five protocol designs, we show how a bit-oriented protocol is actually implemented by using the High-level Data Link Control (HDLC) Protocol as an example. We also discuss a popular byte-oriented protocol, Point-to-Point Protocol (PPP).

FRAMING

Data transmission in the physical layer means moving bits in the form of a signal from the source to the destination. The physical layer provides bit synchronization to ensure that the sender and receiver use the same bit durations and timing. The data link layer, on the other hand, needs to pack bits into frames, so that each frame is distinguishable from another. Our postal system practices a type of framing. The simple act of inserting a letter into an envelope separates one piece of information from another; the envelope serves as the delimiter. In addition, each envelope defines the sender and receiver addresses since the postal system is a many-to-many carrier facility.
Framing in the data link layer separates a message from one source to a destination, or from other messages to other destinations, by adding a sender address and a destination address. The destination address defines where the packet is to go; the sender address helps the recipient acknowledge the receipt. Although the whole message could be packed in one frame, that is not normally done. One reason is that a frame can be very large, making flow and error control very
inefficient. When a message is carried in one very large frame, even a single-bit error would require the retransmission of the whole message. When a message is divided into smaller frames, a single-bit error affects only that small frame.

Fixed-Size Framing
Frames can be of fixed or variable size. In fixed-size framing, there is no need for defining the boundaries of the frames; the size itself can be used as a delimiter. An example of this type of framing is the ATM wide-area network, which uses frames of fixed size called cells.

Variable-Size Framing
Our main discussion in this chapter concerns variable-size framing, prevalent in local area networks. In variable-size framing, we need a way to define the end of the frame and the beginning of the next. Historically, two approaches were used for this purpose: a character-oriented approach and a bit-oriented approach.

Character-Oriented Protocols
In a character-oriented protocol, data to be carded are 8-bit characters from a coding system such as ASCII . The header, which normally carries the source and destination addresses and other control information, and the trailer, which carries error detection or error correction redundant bits, are also multiples of 8 bits. To separate one frame from the next, an 8-bit (1-byte) flag is added at the beginning and the end of a frame. The flag, composed of protocol-dependent special characters, signals the start or end of a frame. Character-oriented framing was popular when only text was exchanged by the data
link layers. The flag could be selected to be any character not used for text communication. Now, however, we send other types of information such as graphs, audio, and video. Any pattern used for the flag could also be part of the information. If this happens, the receiver, when it encounters this pattern in the middle of the data, thinks it has reached the end of the frame. To fix this problem, a byte-stuffing strategy was added to character-oriented framing. In byte stuffing (or character stuffing), a special byte is added to the data section of the frame when there is a character with the same pattern as the flag. The data section is stuffed with an extra byte. This byte is usually called the escape character (ESC), which has a predefined bit pattern. Whenever the receiver encounters the ESC character, it removes it from the data section and treats the next character as data, not a delimiting flag. Byte stuffing by the escape character allows the presence of the flag in the data section of the frame, but it creates another problem. What happens if the text contains one or more escape characters followed by a flag? The receiver removes the escape character, but keeps the flag, which is incorrectly interpreted as the end of the frame. To solve this problem, the escape characters that are part of the text must also be marked by another escape character. In other words, if the escape character is part of the text, an extra one is added to show that the second one is part of the text. Character-oriented protocols present another problem in data communications. The universal coding systems in use today, such as Unicode, have 16-bit and 32-bit characters that conflict with 8-bit characters. We can say that in general, the tendency is moving toward the bit-oriented protocols that we discuss next.


Bit-Oriented Protocols
In a bit-oriented protocol, the data section of a frame is a sequence of bits to be interpreted by the upper layer as text, graphic, audio, video, and so on. However, in addition to headers (and possible trailers), we still need a delimiter to separate one frame from the other. Most protocols use a special 8-bit pattern flag 01111110 as the delimiter to define the beginning and the end of the frame.

This flag can create the same type of problem we saw in the byte-oriented protocols. That is, if the flag pattern appears in the data, we need to somehow inform the receiver that this is not the end of the frame. We do this by stuffing 1 single bit (instead of 1 byte) to prevent the pattern from looking like a flag. The strategy is called bit stuffing. In bit stuffing, if a 0 and five consecutive 1 bits are encountered, an extra 0 is added. This extra stuffed bit is eventually removed from the data by the receiver. Note that the extra bit is added after one 0 followed by five ls regardless of the value of the next bit. This guarantees that the flag field sequence does not inadvertently appear in the frame.

Bit stuffing is the process of adding one extra 0 whenever five consecutive Is follow a 0 in the data, so that the receiver does not mistake the pattern 0111110 for a flag.

FLOW AND ERROR CONTROL
Data communication requires at least two devices working together, one to send and the other to receive. Even such a basic arrangement requires a great deal of coordination for an intelligible exchange to occur. The most important responsibilities of the data link layer are flow control and error control. Collectively, these functions are known as data link control.

Flow Control
Flow control coordinates the amount of data that can be sent before receiving an acknowl- edgment and is one of the most important duties of the data link layer. In most protocols, flow control is a set of procedures that tells the sender how much data it can transmit before it must wait for an acknowledgment from the receiver. The flow of data must not be allowed to overwhelm the receiver. Any receiving device has a limited speed at which it can process incoming data and a limited amount of memory in which to store incoming data. The receiving device must be able to inform the sending device before those limits are reached and to request that the transmitting device send fewer frames or stop temporarily. Incoming data must be checked and processed before they can be used. The rate of such processing is often slower than the rate of transmission. For this reason, each receiving device has a block of memory, called a buffer, reserved for storing incoming data until they are processed. If the buffer begins to fill up, the receiver must be able to tell the sender to halt transmission until it is once again able to receive.

Error Control
Error control is both error detection and error correction. It allows the receiver to inform the sender of any frames lost or damaged in transmission and coordinates the retransmission of those frames by the sender. In the data link layer, the term error control refers primarily to methods of error detection and retransmission. Error control in the data link layer is often implemented simply: Any time an error is detected in an exchange, specified frames are retransmitted. This process is called automatic repeat request (ARQ).

PROTOCOLS
Now let us see how the data link layer can combine flaming, flow control, and error control to achieve the delivery of data from one node to another. The protocols are normally implemented in software by using one of the common programming languages. To make our discussions language-free, we have written in pseudocode a version of each protocol that concentrates mostly on the procedure instead of delving into the details of language rules. We divide the discussion of protocols into those that can be used for noiseless (error-free) channels and those that can be used for noisy (error-creating) channels. The protocols in the first category cannot be used in real life, but they serve as a basis for understanding the protocols of noisy channels. There is a difference between the protocols we discuss here and those used in real networks. All the protocols we discuss are unidirectional in the sense that the data frames travel from one node, called the sender, to another node, called the receiver. Although special frames, called acknowledgment (ACK) and negative acknowledgment (NAK) can flow in the opposite direction for flow and error control purposes, data flow in only one direction. In a real-life network, the data link protocols are implemented as bidirectional; data flow in both directions. In these protocols the flow and error control information such as ACKs and NAKs is included in the data frames in a technique called piggybacking. Because bidirectional protocols are more complex than unidirectional ones, we chose the latter for our discussion. If they are understood, they can be extended to bidirectional protocols. We leave this extension as an exercise.

NOISELESS CHANNELS
Let us first assume we have an ideal channel in which no frames are lost, duplicated, or corrupted. We introduce two protocols for this type of channel. The first is a protocol that does not use flow control; the second is the one that does. Of course, neither has error control because we have assumed that the channel is a perfect noiseless channel.
Simplest Protocol
Our first protocol, which we call the Simplest Protocol for lack of any other name, is one that has no flow or error control. Like other protocols we will discuss in this chapter, it is a unidirectional protocol in which data frames are traveling in only one direction--from the sender to receiver. We assume that the receiver can immediately handle any frame it receives with a processing time that is small enough to be negligible. The data link layer of the receiver immediately removes the header from the frame and hands the data packet to its network layer, which can also accept the packet immediately. In other words, the receiver can never be overwhelmed with incoming frames.

Design
There is no need for flow control in this scheme. The data link layer at the sender site gets data from its network layer, makes a frame out of the data, and sends it. The data link layer at the receiver site receives a frame from its physical layer, extracts data from the frame, and delivers the data to its network layer. The data link layers of the sender and receiver provide transmission services for their network layers. The data link layers use the services provided by their physical layers (such as signaling, multiplexing, and so on) for the physical transmission of bits. We need to elaborate on the procedure used by both data link layers. The sender site cannot send a frame until its network layer has a data packet to send. The receiver site cannot deliver a data packet to its network layer until a frame arrives. If the protocol is implemented as a procedure, we need to introduce the idea of events in the protocol. The procedure at the sender site is constantly running; there is no action until there is a request from the network layer. The procedure at the receiver site is also constantly running, but there is no action until notification from the physical layer arrives. Both procedures are constantly running because they do not know when the corresponding events will occur.

Analysis
The algorithm has an infinite loop, which means lines 3 to 9 are repeated forever once the program starts. The algorithm is an event-driven one, which means that it sleeps (line 3) until an event wakes it up (line 4). This means that there may be an undefined span of time between the execution of line 3 and line 4; there is a gap between these actions. When the event, a request from the network layer, occurs, lines 6 though 8 are executed. The program then repeats the loop and again sleeps at line 3 until the next occurrence of the event. We have written pseudocode for the main process. We do not show any details for the modules GetData, Make-Frame, and SendFrame. GetData() takes a data packet from the network layer, MakeFrame() adds a header and delimiter flags to the data packet to make a frame, and SendFrame() delivers the frame to the physical layer for transmission.

This algorithm has the same format as Algorithm 11.1, except that the direction of the frames and data is upward. The event here is the arrival of a data frame. After the event occurs, the data link layer receives the frame from the physical layer using the ReceiveFrame() process, extracts the data from the frame using the ExtractData() process, and delivers the data to the network layer using the DeliverData() process. Here, we also have an event-driven algorithm because the algorithm never knows when the data frame will arrive.


Stop-and-Wait Protocol
If data frames arrive at the receiver site faster than they can be processed, the frames must be stored until their use. Normally, the receiver does not have enough storage space, especially if it is receiving data from many sources. This may result in either the discarding of frames or denial of service. To prevent the receiver from becoming over- whelmed with frames,we somehow need to tell the sender to slow down. There must be feedback from the receiver to the sender. The protocol we discuss now is called the Stop-and-Wait Protocol because the sender sends one frame, stops until it receives confirmation from the receiver (okay to go ahead), and then sends the next frame. We still have unidirectional communication for data frames, but auxiliary ACK frames (simple tokens of acknowledgment) travel from the other direction. We add flow control to our previous protocol.

NOISY CHANNELS
Although the Stop-and-Wait Protocol gives us an idea of how to add flow control to its predecessor, noiseless channels are nonexistent. We can ignore the error (as we some- times do), or we need to add error control to our protocols. We discuss three protocols in this section that use error control.

Stop-and-Wait Automatic Repeat Request
Our first protocol, called the Stop-and-Wait Automatic Repeat Request (Stop-and- Wait ARQ), adds a simple error control mechanism to the Stop-and-Wait Protocol. Let us see how this protocol detects and corrects errors. To detect and correct corrupted frames, we need to add redundancy bits to our data frame (see Chapter 10). When the frame arrives at the receiver site, it is checked and if it is corrupted, it is silently discarded. The detection of errors in this protocol is mani- fested by the silence of the receiver. Lost frames are more difficult to handle than corrupted ones. In our previous protocols, there was no way to identify a frame. The received frame could be the correct one, or a duplicate, or a frame out of order. The solution is to number the frames. When the receiver receives a data frame that is out of order, this means that frames were either lost or duplicated. The corrupted and lost frames need to be resent in this protocol. If the receiver does not respond when there is an error, how can the sender know which frame to resend? To remedy this problem, the sender keeps a copy of the sent frame. At the same time, it starts a timer. If the timer expires and there is no ACK for the sent frame, the frame is resent, the copy is held, and the timer is restaxted. Since the protocol uses the stop-and-wait mecha- nism, there is only one specific frame that needs an ACK even though several copies of
the same frame can be in the network.

Error correction in Stop-and-Wait ARQ is done by keeping a copy of the sent frameand retransmitting of the frame when the timer expires.

Since ann ACK frame can also be corrupted and lost, it too needs redundancy bits
and a sequence number. The ACK frame for this protocol has a sequence number field.
In this protocol, the sender simply discards a corrupted ACK frame or ignores an
out-of-order one.

Sequence Numbers
As we discussed, the protocol specifies that frames need to be numbered. This is done by using sequence numbers. A field is added to the data frame to hold the sequence number of that frame. One important consideration is the range of the sequence numbers. Since we want to minimize the frame size, we look for the smallest range that provides unambiguous communication.

Selective Repeat Automatic Repeat Request
Go-Back-N ARQ simplifies the process at the receiver site. The receiver keeps track of only one variable, and there is no need to buffer out-of-order frames; they are simply discarded. However, this protocol is very inefficient for a noisy link. In a noisy link a frame has a higher probability of damage, which means the resending of multiple frames. This resending uses up the bandwidth and slows down the transmission. For noisy links, there is another mechanism that does not resend N frames when just one frame is damaged; only the damaged frame is resent. This mechanism is called Selective Repeat ARQ. It is more efficient for noisy links, but the processing at the receiver is more complex. Wigidows The Selective Repeat Protocol also uses two windows: a send window and a receive win- dow. However, there are differences between the windows in this protocol and the ones in Go-Back-N. First, the size of the send window is much smaller; it is 2 m-1. The reason for this will be discussed later. Second, the receive window is the same size as the send window. The send window maximum size can be 2 m-1. For example, if m = 4, the sequence numbers go from 0 to 15, but the size of the window is just 8 (it is 15 in the Go-Back-N Protocol). The smaller window size means less efficiency in filling the pipe, but the fact that there are fewer duplicate frames can compensate for this. The protocol uses the same variables as we discussed for Go-Back-N.


Piggybacking
The three protocols we discussed in this section are all unidirectional: data frames flow in only one direction although control information such as ACK and NAK frames can travel in the other direction. In real life, data frames are normally flowing in both directions: from node A to node B and from node B to node A. This means that the control information also needs to flow in both directions. A technique called piggybacking is used to improve the efficiency of the bidirectional protocols. When a frame is carrying data from A to B, it can also carry control information about arrived (or lost) frames from B; when a frame is carrying data from B to A, it can also carry control information about the arrived (or lost) frames from A. We show the design for a Go-Back-N ARQ .Note that each node now has two windows: one send window and one receive window. Both also need to use a timer. Both are involved in three types of events: request, amval, and time-out. However, the arrival event here is complicated; when a frame arrives, the
site needs to handle control information as well as the frame itself. Both of these concerns must be taken care of in one event, the arrival event. The request event uses only the send window at each site; the arrival event needs to use both windows. An important point about piggybacking is that both sites must use the same algorithm. This algorithm is complicated because it needs to combine two arrival events into one. We leave this task as an exercise.

Telephone networks

TELEPHONE NETWORK

Telephone networks use circuit switching. The telephone network had its beginnings in the late 1800s. The entire network, which is referred to as the plain old telephone system (POTS), was originally an analog system using analog signals to transmit voice. With the advent of the computer era, the network, in the 1980s, began to carry data inaddition to voice. During the last decade, the telephone network has undergone many technical changes. The network is now digital as well as analog.

Major Components
there are three major components these are The telephone network, is made of three major components:

local loops, trunks, and switching offices. The telephone network has several levels of
switching offices such as end offices, tandem offices, and regional offices.

Local Loops
One component of the telephone network is the local loop, a twisted-pair cable that connects the subscriber telephone to the nearest end office or local central office. The local loop, when used for voice, has a bandwidth of 4000 Hz (4 kHz). It is interesting to examine the telephone number associated with each local loop. The first three digits of a local telephone number define the office, and the next four digits define the local loop number.

Trunks
Trunks are transmission media that handle the communication between offices. A trunk normally handles hundreds or thousands of connections through multiplexing. Transmission is usually through optical fibers or satellite links.

Switching Offices
To avoid having a permanent physical link between any two subscribers, the telephone company has switches located in a switching office. A switch connects several local loops or trunks and allows a connection between different subscribers.

LATAs
After the divestiture of 1984 (see Appendix E), the United States was divided into more than 200 local-access transport areas (LATAs). The number of LATAs has increased since then. A LATA can be a small or large metropolitan area. A small state may have one single LATA; a large state may have several LATAs. A LATA boundary may overlap the boundary of a state; part of a LATA can be in one state, part in another state.

Intra-LATA Services
The services offered 'by the common carriers (telephone companies) inside a LATA are called intra-LATA services. The carrier that handles these services is called a local exchange carrier (LEC). Before the Telecommunications Act of 1996 (see Appendix E), intra-LATA services were granted to one single carrier. This was a monopoly. After 1996, more than one carder could provide services inside a LATA. The carder that provided services before 1996 owns the cabling system (local loops) and is called the incumbent local exchange carrier (ILEC). The new carriers that can provide services are called competitive local exchange carriers (CLECs). To avoid the costs of new cabling, it was agreed that the ILECs would continue to provide the main services, and the CLECs would provide other services such as mobile telephone service, toll calls inside a LATA, and so on. Communication inside a LATA is handled by end switches and tandem switches. A call that can be completed by using only end offices is considered toll-free. A call that
has to go through a tandem office (intra-LATA toll office) is charged.Intra-LATA services are provided by local -exchange carriers. Since 1996, there are two types of LECs: incumbent local exchange carriers and competitive local exchange carriers.

Inter-LATA Services

The services between LATAs are handled by interexchange carriers (IXCs). These carders, sometimes called long-distance companies, provide communication services between two customers in different LATAs. After the act of 1996 (see Appendix E), these services can be provided by any carder, including those involved in intra-LATA services. The field is wide open. Carders providing inter-LATA services include AT&T, MCI, WorldCom, Sprint, and Verizon. The IXCs are long-distance carriers that provide general data communications services including telephone service. A telephone call going through an IXC is normally digitized, with the carders using several types of networks to provide service.

Points of Presence
As we discussed, intra-LATA services can be provided by several LECs (one ILEC and possibly more than one CLEC). We also said that inter-LATA services can be provided by several IXCs. How do these carriers interact with one another? The answer is, via a switching office called a point of presence (POP). Each IXC that wants to provide interLATA services in a LATA must have a POP in that LATA. The LECs that provide services inside the LATA must provide connections so that every subscriber can have access to all POPs. Figure 9.3 illustrates the concept. A subscriber who needs to make a connection with another subscriber is connected first to an end switch and then, either directly or through a tandem switch, to a POP. The call now goes from the POP of an IXC (the one the subscriber has chosen) in the source LATA to the POP of the same IXC in the destination LATA. The call is passed through the toll office of the IXC and is carried through the network provided by the IXC.

Signaling
The telephone network, at its beginning, used a circuit-switched network with dedicated links (multiplexing had not yet been invented) to transfer voice communication. As we saw in Chapter 8, a circuit-switched network needs the setup and teardown phases to establish and terminate paths between the two communicating parties. In the beginning, this task was performed by human operators. The operator room was a center to which all subscribers were connected. A subscriber who wished to talk to another subscriber picked up the receiver (off-hook) and rang the operaton The operator, after listening to the caller and getting the identifier of the called party, connected the two by using a wire with two plugs inserted into the corresponding two jacks. A dedicated circuit was created in this way. One of the parties, after the conversation ended, informed the operator to disconnect the circuit. This type of signaling is called in-band signaling because the same circuit can be used for both signaling and voice communication. Later, the signaling system became automatic. Rotary telephones were invented that sent a digital signal defining each digit in a multidigit telephone number. The switches in the telephone companies used the digital signals to create a connection between the caller and the called parties. Both in-band and out-of-band signaling were used. In in-band signaling, the 4-kHz voice channel was also used to provide signaling. In out-of-band signaling, a portion of the voice channel bandwidth was used for signaling; the voice bandwidth and the signaling bandwidth were separate. As telephone networks evolved into a complex network, the functionality of the signaling system increased. The signaling system was required to perform other tasks such as

1. Providing dial tone, ring tone, and busy tone
2. Transferring telephone numbers between offices
3. Maintaining and monitoring the call
4. Keeping billing information
5. Maintaining and monitoring the status of the telephone network equipment
6. Providing other functions such as caller ID, voice mail, and so on These complex tasks resulted in the provision of a separate network for signaling. This means that a telephone network today can be thought of as two networks: a signaling network and a data transfer network. The tasks of data transfer and signaling are separated in modern telephone networks: data transfer is done by one network, signaling by another.

However, we need to emphasize a point here. Although the two networks are separate, this does not mean that there are separate physical links everywhere; the two networks may use separate channels of the same link in parts of the system. Data Transfer Network The data transfer network that can carry multimedia information today is, for the most part, a circuit-switched network, although it can also be a packet-switched network. This network follows the same type of protocols and model as other networks discussed in this book.

Signaling Network
The signaling network, which is our main concern in this section, is a packet-switched network involving the layers similar to those in the OSI model or Internet model, nature of signaling makes it more suited to a packet-switching network with different layers. For example, the information needed to convey a telephone address can easily be encapsulated in a packet with all the error control and addressing information.

Signaling System Seven (SS7)
The protocol that is used in the signaling network is called Signaling System Seven (SS7). It is very similar to the five-layer Internet model

Physical Layer: MTP Level 1 The physical layer in SS7 called message transport part (MTP) level 1 uses several physical layer specifications such as T-1 (1.544 Mbps) and DC0 (64 kbps).

Data Link Layer: MTP Level 2 The MTP level 2 layer provides typical data link layer services such as packetizing, using source and destination address in the packet header, and CRC for error checking.

Network Layer: MTP Level 3 The MTP level 3 layer provides end-to-end connectivity by using the datagram approach to switching. Routers and switches route the signal packets from the source to the destination.

Transport Layer: SCCP The signaling connection control point (SCCP) is used for special services such as 800-call processing.

Upper Layers: TUP, TCAP, and ISUP There are three protocols at the upper layers. Telephone user port (TUP) is responsible for setting up voice calls. It receives the dialed digits and routes the calls. Transaction capabilities application port (TCAP) provides remote calls that let an application program on a computer invoke a procedure on another computer. ISDN user port (ISUP) can replace TUP to provide services similar to those of an ISDN network.

Services Provided by Telephone Networks
Telephone companies provide two types of services: analog and digital. Analog Services In the beginning, telephone companies provided their subscribers with analog services. These services still continue today. We can categorize these services as either analog switched services or analog leased services. Analog Switched Services This is the familiar dial-up service most often encountered when a home telephone is used. The signal on a local loop is analog, and the bandwidth is usually between 0 and 4000 Hz. A local call service is normally provided for a flat monthly rate, although in some LATAs, the carder charges for each call or a set of calls. The rationale for a non flat-rate charge is to provide cheaper service for those customers who do not make many calls. A toll call can be intra-LATA or inter-LATA. If the LATA is geographically large, a call may go through a tandem office (toll office) and the subscriber will pay a fee for the call. The inter-LATA calls are long-distance calls and are charged as such. Another service is called 800 service. If a subscriber (normally an organization) needs to provide free connections for other subscribers (normally customers), it can request the 800 service. In this case, the call is free for the caller, but it is paid by the callee. An organization uses this service to encourage customers to call. The rate is less expensive than that for a normal long-distance call. The wide-area telephone service (WATS) is the opposite of the 800 service. The latter are inbound calls paid by the organization; the former are outbound calls paid by the organization. This service is a less expensive alternative to regular toll calls; charges are based on the number of calls. The service can be specified as outbound calls to the same state, to several states, or to the whole country, with rates charged accordingly. The 900 services are like the 800 service, in that they are inbound calls to a sub- scriber. However, unlike the 800 service, the call is paid by the caller and is normally much more expensive than a normal long-distance call. The reason is that the carrier charges two fees: the first is the long-distance toll, and the second is the fee paid to the callee for each call.

Analog Leased Service
An analog leased service offers customers the opportunity to lease a line, sometimes called a dedicated line, that is permanently connected toanother customer. Although the connection still passes through the switches in the telephone network, subscribers experience it as a single line because the switch is always closed; no dialing is needed.

Digital Services
Recently telephone companies began offering digital services to their subscribers. Digital services are less sensitive than analog services to noise and other forms of interference. The two most common digital services axe switched/56 service and digital data service (DDS). Switched/56 Service Switched/56 service is the digital version of an analog switched line. It is a switched digital service that allows data rates of up to 56 kbps. To communivative through this service, both parties must subscribe. A caller with normal telephone service cannot connect to a telephone or computer with switched/56 service even if the caller is using a modem. On the whole, digital and analog services represent two completely different domains for the telephone companies. Because the line in a switched/ 56 service is already digital, subscribers do not need modems to transmit digital data. However, they do need another device called a digital service unit (DSU). Digital Data Service Digital data service (DDS) is the digital version of an analog
leased line; it is a digital leased line with a maximum data rate of 64 kbps.

Datagram networks

In data communications, we need to send messages from one end system to another. If the message is going to pass through a packet-switched network, it needs to be divided into packets of fixed or variable size. The size of the packet is determined by the network and the governing protocol. In packet switching, there is no resource allocation for a packet. This means that there is no reserved bandwidth on the links, and there is no scheduled processing time for each packet. Resources are allocated on demand. The allocation is done on a first- come, first-served basis. When a switch receives a packet, no matter what is the source or destination, the packet must wait if there are other packets being processed. As with other systems in our daily life, this lack of reservation may create delay. For example, if we do not have a reservation at a restaurant, we might have to wait.

In a packet-switched network, there is no resource reservation;
resources are allocated on demand.

In a datagram network, each packet is treated independently of all others. Even if a packet is part of a multipacket transmission, the network treats it as though it existed alone. Packets in this approach are referred to as datagrams. Datagram switching is normally done at the network layer. We briefly discuss datagram networks here as a comparison with circuit-switched and virtual-circuit- switched networks. The datagram networks are sometimes referred to as connectionless networks. The term connectionless here means that the switch (packet switch) does not keep information about the connection state. There are no setup or teardown phases. Each packet is treated the same by a switch regardless of its source or destination.

Routing Table

If there are no setup or teardown phases, how are the packets routed to their destinations in a datagram network? In this type of network, each switch (or packet switch) has a routing table which is based on the destination address. The routing tables are dynamic and are updated periodically. The destination addresses and the corresponding forwarding output ports are recorded in the tables. This is different from the table of a circuit- switched network in which each entry is created when the setup phase is completed and deleted when the teardown phase is over.

Destination Address

Every packet in a datagram network carries a header that contains, among other information, the destination address of the packet. When the switch receives the packet, this destination address is examined; the routing table is consulted to find the corresponding port through which the packet should be forwarded. This address, unlike the address in a virtual-circuit-switched network, remains the same during the entire journey of the packet.

The destination address in the header of a acket in a datagram network
remains the same during the entire journey of the packet.

Efficiency
The efficiency of a datagram network is better than that of a circuit-switched network; resources are allocated only when there are packets to be transferred. If a source sends a packet and there is a delay of a few minutes before another packet can be sent, the resources can be reallocated during these minutes for other packets from other sources.


Delay
There may be greater delay in a datagram network than in a virtual-circuit network. Although there are no setup and teardown phases, each packet may experience a wait at a switch before it is forwarded. In addition, since not all packets in a message necessarily travel through the same switches, the delay is not uniform for the packets of a message.


Datagram Networks in the Internet
As we will see in future chapters, the Internet has chosen the datagram approach to switching at the network layer. It uses the universal addresses defined in the network layer to route packets from the source to the destination.

Switching in the Lnternet is done by using the datagram
approach to packet switching at the network layer.

VIRTUAL-CIRCUIT NETWORKS
A virtual-circuit network is a cross between a circuit-switched network and a datagram network. It has some characteristics of both.

1. As in a circuit-switched network, there are setup and teardown phases in addition to the data transfer phase.
2. Resources can be allocated during the setup phase, as in a circuit-switched network, or on demand, as in a datagram network.
3. As in a datagram network, data are packetized and each packet carries an address in the header. However, the address in the header has local jurisdiction (it defines what should be the next switch and the channel on which the packet is being carried), not end-to-end jurisdiction. The reader may ask how the intermediate switches know where to send the packet if there is no final destination address carried by a packet. The answer will be clear when we discuss virtual-circuit identifiers in the next section.
4. As in a circuit-switched network, all packets follow the same path established during the connection.
5. A virtual-circuit network is normally implemented in the data link layer, while a circuit-switched network is implemented in the physical layer and a datagram network in the network layer. But this may change in the future.


Addressing
In a virtual-circuit network, two types of addressing are involved: global and local (virtual-circuit identifier).

Global Addressing
A source or a destination needs to have a global address--an address that can be unique in the scope of the network or internationally if the network is part of an international network. However, we will see that a global address in virtual-circuit networks is used only to create a virtual-circuit identifier, as discussed next.

Virtual- Circuit Identifier
The identifier that is actually used for data transfer is called the virtual-circuit identifier (VCI). A VCI, unlike a global address, is a small number that has only switch scope; it is used by a frame between two switches. When a frame arrives at a switch, it has a VCI; when it leaves, it has a different VCI.

Data Transfer Phase
To transfer a frame from a source to its destination, all switches need to have a table entry for this virtual circuit. The table, in its simplest form, has four columns. This means that the switch holds four pieces of information for each virtual circuit that is already set up. We show later how the switches make their table entries, but for the moment we assume that each switch has a table with entries for all active virtual circuits. The data transfer phase is active until the source sends all its frames to the destination. The procedure at the switch is the same for each frame of a message. The process creates a virtual circuit, not a real circuit, between the source and destination.

Setup Phase
In the setup phase, a switch creates an entry for a virtual circuit. For example, suppose source A needs to create a virtual circuit to B. Two steps are required: the setup request and the acknowledgment.

Teardown Phase
In this phase, source A, after sending all frames to B, sends a special frame called a teardown request. Destination B responds with a teardown confirmation frame. All switches delete the corresponding entry from their tables.


Efficiency
As we said before, resource reservation in a virtual-circuit network can be made during the setup or can be on demand during the data transfer phase. In the first case, the delay for each packet is the same; in the second case, each packet may encounter different delays. There is one big advantage in a virtual-circuit network even if resource allocation is on demand. The source can check the availability of the resources, without actually reserving it. Consider a family that wants to dine at a restaurant. Although the restaurant may not accept reservations (allocation of the tables is on demand), the family can call and find out the waiting time. This can save the family time and effort.

In virtual-circuit switching, all packets belonging to the same source and destination travel the same path; but the packets may arrive at the destination
with different delays if resource allocation is on demand.

Delay in Virtual-Circuit Networks
In a virtual-circuit network, there is a one-time delay for setup and a one-time delay for teardown. If resources are allocated during the setup phase, there is no wait time for individual packets.

Switching at the data link layer in a switched WAN is normally
implemented by using virtual-circuit techniques.

Circuit Switched Network

A network is a set of connected devices. Whenever we have multiple devices, we have the problem of how to connect them to make one-to-one communication possible. One solution is to make a point-to-point connection between each pair of devices (a mesh topology) or between a central device and every other device (a star topology). These methods, however, are impractical and wasteful when applied to very large networks. The number and length of the links require too much infrastructure to be cost-efficient, and the majority of those links would be idle most of the time. Other topologies employing multipoint connections, such as a bus, are ruled out because the distances between devices and the total number of devices increase beyond the capacities of the media and equipment. A better solution is switching. A switched network consists of a series of interlinked nodes, called switches. Switches are devices capable of creating temporary connections between two or more devices linked to the switch. In a switched network, some of these nodes are connected to the end systems (computers or telephones, for example). Others are used only for routing.

Traditionally, three methods of switching have been important: circuit switching, packet switching, and message switching. The first two are commonly used today. The third has been phased out in general communications but still has networking applications. We can then divide today's networks into three broad categories: circuit-switched networks, packet-switched networks, and message-switched. Packet-switched networks can further be divided into two subcategories--virtual-circuit networks and datagram networks.

We can say that the virtual-circuit networks have some common characteristics with circuit-switched and datagram networks. Thus, we first discuss circuit-switched networks, then datagram networks, and finally virtual-circuit networks. Today the tendency in packet switching is to combine datagram networks and virtual- circuit networks. Networks route the first packet based on the datagram addressing idea, but then create a virtual-circuit network for the rest of the packets coming from the same source and going to the same destination. In message switching, each switch stores the whole message and forwards it to the next switch. Although, we don't see message switching at lower layers, it is still used in some applications like electronic mail (e-mail).

CIRCUIT-SWITCHED NETWORKS

A circuit-switched network consists of a set of switches connected by physical links. A connection between two stations is a dedicated path made of one or more links. How- ever, each connection uses only one dedicated channel on each link. Each link is normally divided into n channels by using FDM or TDM.

A circuit-switched network is made of a set of switches connected by physical links,in which each link is divided into n channels.

We have explicitly shown the multiplexing symbols to emphasize the division of the link into channels even though multiplexing can be implicitly included in the switch fabric. The end systems, such as computers or telephones, are directly connected to a switch. We have shown only two end systems for simplicity. When end system A needs to communicate with end system M, system A needs to request a connection to M that must be accepted by all switches as well as by M itself. This is called the setup phase; a circuit (channel) is reserved on each link, and the combination of circuits or channels defines the dedicated path. After the dedicated path made of connected circuits (channels) is established, data transfer can take place. After all data have been transferred, the circuits are tom down.
We need to emphasize several points here:

1. switching takes place at the physical layer.

2. Before starting communication, the stations must make a reservation for the resources to be used during the communication. These resources, such as channels (bandwidth in FDM and time slots in TDM), switch buffers, switch processing time, and switch input/output ports, must remain dedicated during the entire duration of data transfer until the teardown phase.

3. transferred between the two stations axe not packetized (physical layer transfer of the signal). The data are a continuous flow sent by the source station and received by the destination station, although there may be periods of silence.

4. There is no addressing involved during data transfer. The switches route the data based on their occupied band (FDM) or time slot (TDM). Of course, there is end-to- end addressing used during the setup phase, as we will see shortly.

In circuit switching, the resources need to be reserved during the setup phase; the resources remain dedicated for the entire duration of data transfer until the teardown phase.

Three Phases

The actual communication in a circuit-switched network requires three phases: connection setup, data transfer, and connection teardown.
Setup Phase

Before the two parties (or multiple parties in a conference call) can communicate, a dedicated circuit (combination of channels in links) needs to be established. The end systems are normally connected through dedicated lines to the switches, so connection setup means creating dedicated channels between the switches.

Data Transfer Phase
After the establishment of the dedicated circuit (channels), the two parties can transfer data.

Teardown Phase

When one of the parties needs to disconnect, a signal is sent to each switch to release the resources.

Efficiency
It can be argued that circuit-switched networks are not as efficient as the other two types of networks because resources are allocated during the entire duration of the connection. These resources are unavailable to other connections. In a telephone network, people normally terminate the communication when they have finished their conversation. However, in computer networks, a computer can be connected to another computer even if there is no activity for a long time. In this case, allowing resources to be dedicated means that other connections are deprived.

Delay

Although a circuit-switched network normally has low efficiency, the delay in this type of network is minimal. During data transfer the data are not delayed at each switch; the resources are allocated for the duration of the connection.

Circuit-Switched Technology in Telephone Networks

the telephone companies have previously chosen the circuit switched approach to switching in the physical layer; today the tendency is moving toward other switching techniques. For example, the telephone number is used as the global address, and a signaling system (called SS7) is used for the setup and teardown phases.

Unguided Tr. Me

Unguided Transmission Media

Unguided transmission media is data signals that flow through the air. They are not guided or bound to a channel to follow. They are classified by the type of wave propagation.

RF Propagation

There are three types of RF (radio frequency) propagation:

• Ground Wave

• Ionospheric

• Line of Sight (LOS)

Ground wave propagation follows the curvature of the Earth. Ground waves have carrier frequencies up to 2 MHz. AM radio is an example of ground wave propagation.

Ionospheric propagation bounces off of the Earth's ionospheric layer in the upper atmosphere. It is sometimes called double hop propagation. It operates in the frequency range of 30 - 85 MHz. Because it depends on the Earth's ionosphere, it changes with the weather and time of day. The signal bounces off of the ionosphere and back to earth. Ham radios operate in this range.

Line of sight propagation transmits exactly in the line of sight. The receive station must be in the view of the transmit station. It is sometimes called space waves or tropospheric propagation. It is limited by the curvature of the Earth for ground-based stations (100 km, from horizon to horizon). Reflected waves can cause problems. Examples of line of sight propagation are: FM radio, microwave and satellite.

Radio Frequencies

The frequency spectrum operates from 0 Hz (DC) to gamma rays (1019 Hz).

Name	Frequency (Hertz)	Examples
Gamma Rays	1019+
X-Rays	1017
Ultra-Violet Light	7.5 x 1015
Visible Light	4.3 x 1014
Infrared Light	3 x 1011
EHF - Extremely High Frequencies	30 GHz (Giga = 109)	Radar
SHF - Super High Frequencies	3 GHz	Satellite & Microwaves
UHF - Ultra High Frequencies	300 MHz (Mega = 106)	UHF TV (Ch. 14-83)
VHF - Very High Frequencies	30 MHz	FM & TV (Ch2 - 13)
HF - High Frequencies	3 MHz2	Short Wave Radio
MF - Medium Frequencies	300 kHz (kilo = 103)	AM Radio
LF - Low Frequencies	30 kHz	Navigation
VLF - Very Low Frequencies	3 kHz	Submarine Communications
VF - Voice Frequencies	300 Hz	Audio
ELF - Extremely Low Frequencies	30 Hz	Power Transmission

Radio frequencies are in the range of 300 kHz to 10 GHz. We are seeing an emerging technology called wireless LANs. Some use radio frequencies to connect the workstations together, some use infrared technology.

Microwave

Microwave transmission is line of sight transmission. The transmit station must be in visible contact with the receive station. This sets a limit on the distance between stations depending on the local geography. Typically the line of sight due to the Earth's curvature is only 50 km to the horizon! Repeater stations must be placed so the data signal can hop, skip and jump across the country.

Microwaves operate at high operating frequencies of 3 to 10 GHz. This allows them to carry large quantities of data due to their large bandwidth.

Advantages:

1. They require no right of way acquisition between towers.

2. They can carry high quantities of information due to their high operating frequencies.

3. Low cost land purchase: each tower occupies only a small area.

4. High frequency/short wavelength signals require small antennae.

Disadvantages:

1. Attenuation by solid objects: birds, rain, snow and fog.

2. Reflected from flat surfaces like water and metal.

3. Diffracted (split) around solid objects.

4. Refracted by atmosphere, thus causing beam to be projected away from receiver.

Satellite

Satellites are transponders (units that receive on one frequency and retransmit on another) that are set in geostationary orbits directly over the equator. These geostationary orbits are 36,000 km from the Earth's surface. At this point, the gravitational pull of the Earth and the centrifugal force of Earth's rotation are balanced and cancel each other out. Centrifugal force is the rotational force placed on the satellite that wants to fling it out into space.

The uplink is the transmitter of data to the satellite. The downlink is the receiver of data. Uplinks and downlinks are also called Earth stations because they are located on the Earth. The footprint is the "shadow" that the satellite can transmit to, the shadow being the area that can receive the satellite's transmitted signal.

Iridium Telecom System

The Iridium Telecom System is a new satellite system that will be the largest private aerospace project. It is a mobile telecom system intended to compete with cellular phones. It relies on satellites in lower Earth orbit (LEO). The satellites will orbit at an altitude of 900 - 10,000 km in a polar, non-stationary orbit. Sixty-six satellites are planned. The user's handset will require less power and will be cheaper than cellular phones. There will be 100% coverage of the Earth.

Unfortunately, although the Iridium project was planned for 1996-1998, with 1.5 million subscribers by end of the decade, at the time of this writing, it looked very financially unstable.

Guided Tr. Me

A transmission medium can be broadly defined as anything that can carry infor- mation from a source to a destination. For example, the transmission medium for twopeople having a dinner conversation is the air. The air can also be used to convey themessage in a smoke signal or semaphore. For a written message, the transmission medium might be a mail carrier, a truck, or an airplane.In data communications the definition of the information and the transmission medium is more specific. The transmission medium is usually free space, metallic cable,or fiber-optic cable. The information is usually a signal that is the result of a conversionof data from another form.The use of long-distance communication using electric signals started with theinvention of the telegraph by Morse in the 19th century. Communication by telegraphwas slow and dependent on a metallic medium.Extending the range of the human voice became possible when the telephone wasinvented in 1869. Telephone communication at that time also needed a metallic medium to carry the electric signals that were the result of a conversion from the human voice. The communication was, however, unreliable due to the poor quality of the wires. The lines were often noisy and the technology was unsophisticated. Wireless communication started in 1895 when Hertz was able to send high- frequency signals. Later, Marconi devised a method to send telegraph-type messages over the Atlantic Ocean. We have come a long way. Better metallic media have been invented (twisted- pair and coaxial cables, for example). The use of optical fibers has increased the data rate incredibly. Free space (air, vacuum, and water) is used more efficiently, in part due to the technologies (such as modulation and multiplexing). computers and other telecommunication devices use signals to represent data. These signals are transmitted from one device to another in the form of electromagnetic energy, which is propagated through transmission media. Electromagnetic energy, a combination of electric and magnetic fields vibrating in relation to each other, includes power, radio waves, infrared light, visible light, ultraviolet light, and X, gamma, and cosmic rays. Each of these constitutes a portion of the electro- magnetic spectrum. Not all portions of the spectrum are currently usable for telecommu- nications, however. The media to harness those that are usable are also limited to a few types. In telecommunications, transmission media can be divided into two broad catego- ries: guided and unguided. Guided media include twisted-pair cable, coaxial cable, and fiber-optic cable. Unguided medium is free space.

GUIDED MEDIA
Guided media, which are those that provide a conduit from one device to another, include twisted-pair cable, coaxial cable, and fiber-optic cable. A signal traveling along any of these media is directed and contained by the physical limits of the medium. Twisted-pair and coaxial cable use metallic (copper) conductors that accept and transport signals in the form of electric current. Optical fiber is a cable that accepts and transports signals in the form of light.

Twisted-Pair Cable
A twisted pair consists of two conductors (normally copper), each with its own plastic insulation, twisted together, One of the wires is used to carry signals to the receiver, and the other is used only as a ground reference. The receiver uses the difference between the two. In addition to the signal sent by the sender on one of the wires, interference (noise) and crosstalk may affect both wires and create unwanted signals. If the two wires are parallel, the effect of these unwanted signals is not the same in both wires because they are at different locations relative to the noise or crosstalk sources (e.g., one is closer and the other is farther). This results in a difference at the receiver. By twist!ng the pairs, a balance is maintained. For example, suppose in one twist, one wire is closer to the noise source and the other is farther; in the next twist, the reverse is true. Twisting makes it probable that both wires are equally affected by external influences (noise or crosstalk). This means that the receiver, which calculates the difference between the two, receives no unwanted signals. The unwanted signals are mostly canceled out. From the above discussion, it is clear that the number of twists per unit of length (e.g., inch) has some effect on the quality of the cable. Unshielded Versus Shielded Twisted-Pair Cable The most common twisted-pair cable used in communications is referred to as unshielded twisted-pair (UTP). IBM has also produced a version of twisted-pair cable for its use called shielded twisted-pair (STP). STP cable has a metal foil or braided- mesh covering that encases each pair of insulated conductors. Although metal casing improves the quality of cable by preventing the penetration of noise or crosstalk, it is bulkier and more expensive.

Applications
Twisted-pair cables are used in telephone lines to provide voice and data channels. The local loop--the line that connects subscribers to the central telephone office---commonly consists of unshielded twisted-pair cables. The DSL lines that are used by the telephone companies to provide high-data-rate connections also use the high-bandwidth capability of unshielded twisted-pair cables.

Coaxial Cable
Coaxial cable (or coax) carries signals of higher frequency ranges than those in twisted- pair cable, in part because the two media are constructed quite differently. Instead of having two wires, coax has a central core conductor of solid or stranded wire (usually copper) enclosed in an insulating sheath, which is, in turn, encased in an outer conductor of metal foil, braid, or a combination of the two. The outer metallic wrapping serves both as a shield against noise and as the second conductor, which completes the circuit. This outer conductor is also enclosed in an insulating sheath, and the whole cable is protected by a plastic cover. Coaxial Cable Standards Coaxial cables are categorized by their radio government (RG) ratings. Each RG num- ber denotes a unique set of physical specifications, including the wire gauge of the inner conductor, the thickness and type of the inner insulator, the construction of the shield, and the size and type of the outer casing. Each cable defined by an RG rating is adapted for a specialized function.

Coaxial Cable Connectors
To connect coaxial cable to devices, we need coaxial connectors. The most common type of connector used today is the Bayone-Neill-Concelman (BNC), connector.The BNC connector is used to connect the end of the cable to a device, such as a TV set. The BNC T connector is used in Ethernet networks branch out to a connection to a computer or other device. The BNC terminator is used at the end of the cable to prevent the reflection of the signal.

Applications
Coaxial cable was widely used in analog telephone networks where a single coaxial network could carry 10,000 voice signals. Later it was used in digital telephone networks where a single coaxial cable could carry digital data up to 600 Mbps. However, coaxial cable in telephone networks has largely been replaced today with fiber-optic cable. Cable TV networks also use coaxial cables. In the traditional cable TV network, the entire network used coaxial cable. Later, however, cable TV providers replaced most of the media with fiber-optic cable; hybrid networks use coaxial cable only at the network boundaries, near the consumer premises. Cable TV uses RG-59 coaxial cable. Another common application of coaxial cable is in traditional Ethernet LANs Because of its high bandwidth, and consequently high data rate, coaxial cable was chosen for digital transmission in early Ethernet LANs. The 10Base-2, or Thin Ethernet, uses RG-58 coaxial cable with BNC connectors to transmit data at 10 Mbps with a range of 185 m. The 10Base5, or Thick Ethernet, uses RG-11 (thick coaxial cable) to transmit 10 Mbps with a range of 5000 m. Thick Ethernet has specialized connectors.

Fiber-Optic Cable
A fiber-optic cable is made of glass or plastic and transmits signals in the form of light. To understand optical fiber, we first need to explore several aspects of the nature of light. Light travels in a straight line as long as it is moving through a single uniform substance. If a ray of light traveling through one substance suddenly enters another substance (of a different density), the ray changes direction.

Propagation Modes
Current technology supports two modes (multimode and single mode) for propagating light along optical channels, each requiring fiber with different physical characteristics. Multi- mode can be implemented in two forms: step-index or graded-index Multimode Multimode is so named because multiple beams from a light source move through the core in different paths. How these beams move within the cable depends on the structure of the core, In multimode step-index fiber, the density of the core remains constant from the center to the edges. A beam of light moves through this constant density in a straight line until it reaches the interface of the core and the cladding. At the interface, there is an abrupt change due to a lower density; this alters the angle of the beam's motion. The term step index refers to the suddenness of this change, which contributes to the distor- tion of the signal as it passes through the fiber. A second type of fiber, called multimode graded-index fiber, decreases this distor- tion of the signal through the cable. The word index here refers to the index of refraction. As we saw above, the index of refraction is related to density. A graded-index fiber, therefore, is one with varying densities. Density is highest at the center of the core and decreases gradually to its lowest at the edge. Single-Mode Single-mode uses step-index fiber and a highly focused source of light that limits beams to a small range of angles, all close to the horizontal. The single- mode fiber itself is manufactured with a much smaller diameter than that of multimode fiber, and with substantially lower density (index of refraction). The decrease in density results in a critical angle that is close enough to 90 degree to make the propagation of beams almost horizontal. In this case, propagation of different beams is almost identical, and delays are negligible. All the beams arrive at the destination "together" and can be recombined with little distortion to the signal

Fiber Sizes
Optical fibers are defined by the ratio of the diameter of their core to the diameter of their cladding, both expressed in micrometers.

Cable Composition
composition of a typical fiber-optic cable. The outer jacket is made of either PVC or Teflon. Inside the jacket are Kevlar strands to strengthen the cable. Kevlar is a strong material used in the fabrication of bulletproof vests. Below the Kevlar is another plastic coating to cushion the fiber. The fiber is at the center of the cable, and it consists of cladding and core.

Perfortnance
The plot of attenuation versus wavelength in Figure 7.16 shows a very interesting phenomenon in fiber-optic cable. Attenuation is flatter than in the case of twisted-pair cable and coaxial cable. The performance is such that we need fewer (actually 10 times less) repeaters when we use fiber-optic cable.

Applications
Fiber-optic cable is often found in backbone networks because its wide bandwidth is cost-effective. Today, with wavelength-division multiplexing (WDM), we can transfer data at a rate of 1600 Gbps. The SONET network that we discuss in Chapter 17 provides such a backbone. Some cable TV companies use a combination of optical fiber and coaxial cable, thus creating a hybrid network. Optical fiber provides the backbone structure while coaxial cable provides the connection to the user premises. This is a cost-effective con- figuration since the narrow bandwidth requirement at the user end does not justify the use of optical fiber. Local-area networks such as 100Base-FX network (Fast Ethernet) and 1000Base-X also use fiber-optic cable.

Advantages and Disadvantages of Optical Fiber

Advantages Fiber-optic cable has several advantages over metallic cable (twisted-
pair or coaxial).

##Higher bandwidth. Fiber-optic cable can support dramatically higher bandwidths (and hence data rates) than either twisted-pair or coaxial cable. Currently, data rates and bandwidth utilization over fiber-optic cable are limited not by the medium but by the signal generation and reception technology available.

##signal attenuation. Fiber-optic transmission distance is significantly greater than that of other guided media. A signal can mn for 50 km without requiring regeneration. We need repeaters every 5 km for coaxial or twisted-pair cable.

##Immunity to electromagnetic interference. Electromagnetic noise cannot affect fiber-optic cables.

##Resistance to corrosive materials. Glass is more resistant to corrosive materials than copper.

Light weight. Fiber-optic cables are much lighter than copper cables.

Greater immunity to tapping. Fiber-optic cables are more immune to tapping than copper cables. Copper cables create antenna effects that can easily be tapped.

Disadvantages There are some disadvantages in the use of optical fiber.

Installation and maintenance. Fiber-optic cable is a relatively new technology. Its installation and maintenance require expertise that is not yet available everywhere.

Unidirectional light propagation. Propagation of light is unidirectional. If we need bidirectional communication, two fibers are needed.

Cost. The cable and the interfaces are relatively more expensive than those of other guided media. If the demand for bandwidth is not high, often the use of optical fiber cannot be justified.

Multiplexing and spreading

In real life, we have links with limited bandwidths. The wise use of these bandwidths has been, and will be, one of the main challenges of electronic communications. How-ever, he meaning of wise may depend on the application. Sometimes we need to combine several low-bandwidth channels to make use of one channel with a larger bandwidth. Sometimes we need to expand the bandwidth of a channel to achieve goals such as privacy and antijamming. In this chapter, we explore these two broad categories of bandwidth utilization: multiplexing and spreading. In multiplexing, our goal is effi-ciency; we combine several channels into one. In spreading, our goals are privacy and antijamming; we expand the bandwidth of a channel to insert redundancy, which is necessary to achieve these goals.

Bandwidth utilization is the wise use of available bandwidth to achieve specific goals. Efficiency can be achieved by multiplexing; privacy and antijamming can be achieved by spreading.

MULTIPLEXING

Whenever the bandwidth of a medium linking two devices is greater than the band- width needs of the devices, the link can be shared. Multiplexing is the set of techniques that allows the simultaneous transmission of multiple signals across a single data link. As data and telecommunications use increases, so does traffic. We can accommodate this increase by continuing to add individual links each time a new channel is needed; or we can install higher-bandwidth links and use each to carry multiple signals. Today's technology includes high-bandwidth media such as optical fiber and terrestrial and satellite microwaves. Each has a bandwidth far in excess of that needed for the average transmission signal. If the bandwidth of a link is greater than the bandwidth needs of the devices connected to it, the bandwidth is wasted. An efficient system maximizes the utilization of all resources; bandwidth is one of the most precious resources we have in data communication.

There are three basic multiplexing techniques: frequency-division multiplexing, wavelength-division multiplexing, and time-division multiplexing. The first two are techniques designed for analog signals, the third, for digital signals

Frequency-Division Multiplexing
Frequency-division multiplexing (FDM) is an analog technique that can be applied when the bandwidth of a link (in hertz) is greater than the combined bandwidths of the signals to be transmitted. In FDM, signals generated by each sending device modu- late different carder frequencies. These modulated signals are then combined into a single composite signal that can be transported by the link. Carder frequencies are separated by sufficient bandwidth to accommodate the modulated signal. These bandwidth ranges are the channels through which the various signals travel. Channels can be separated by strips of unused bandwidth--guard bands--to prevent signals from overlapping. In addition, carrier frequencies must not interfere with the original data frequencies.

Demultiplexing Process

The demultiplexer uses a series of filters to decompose the multiplexed signal into its constituent component signals. The individual signals are then passed to a demodulator that separates them from their carriers and passes them to the output lines.

The Analog Carrier System

To maximize the efficiency of their infrastructure, telephone companies have tradition- ally multiplexed signals from lower-bandwidth lines onto higher-bandwidth lines. In this way, many switched or leased lines can be combined into fewer but bigger channels. For analog lines, FDM is used. In this analog hierarchy, 12 voice channels are multiplexed onto a higher-bandwidth line to create a group. A group has 48 kHz of bandwidth and supports 12 voice channels. At the next level, up to five groups can be multiplexed to create a composite signal called a supergroup. A supergroup has a bandwidth of 240 kHz and supports up to 60 voice channels. Supergroups can be made up of either five groups or 60 independent voice channels. At the next level, 10 supergroups are multiplexed to create a master group. A master group must have 2.40 MHz of bandwidth, but the need for guard bands between the supergroups increases the necessary bandwidth to 2.52 MHz. Master groups support up to 600 voice channels. Finally, six master groups can be combined into a jumbo group. A jumbo group must have 15.12 MHz (6 x 2.52 MHz) but is augmented to 16.984 MHz to allow for guard bands between the master groups.

Other Applications of FDM

A very common application of FDM is AM and FM radio broadcasting. Radio uses the air as the transmission medium. A special band from 530 to 1700 kHz is assigned to AM radio. All radio stations need to share this band. As discussed in Chapter 5, each AM sta- tion needs 10 kHz of bandwidth. Each station uses a different carrier frequency, which means it is shifting its signal and multiplexing. The signal that goes to the air is a combi- nation of signals. A receiver raceives all these signals, but filters (by tuning) only the one which is desired. Without multiplexing, only one AM station could broadcast to the com- mon link, the air. However, we need to know that there is physical multiplexer or demulti- plexer here. As we will see in Chapter 12 multiplexing is done at the data link layer. The situation is similar in FM broadcasting. However, FM has a wider band of 88 to 108 MHz because each station needs a bandwidth of 200 kHz. Another common use of FDM is in television broadcasting. Each TV channel has its own bandwidth of 6 MHz. The first generation of cellular telephones (still in operation) also uses FDM. Each user is assigned two 30-kHz channels, one for sending voice and the other for receiving. The voice signal, which has a bandwidth of 3 kHz (from 300 to 3300 Hz), is modulated by using FM. Remember that an FM signal has a bandwidth 10 times that of the modulating signal, which means each channel has 30 kHz (10 x 3) of bandwidth. Therefore, each user is given, by the base station, a 60-kHz bandwidth in a range available at the time of the call.

lnplementation

FDM can be implemented very easily. In many cases, such as radio and television broadcasting, there is no need for a physical multiplexer or demultiplexer. As long as the stations agree to send their broadcasts to the air using different carrier frequencies, multiplexing is achieved. In other cases, such as the cellular telephone system, a base station needs to assign a carrier frequency to the telephone user. There is not enough bandwidth in a cell to permanently assign a bandwidth range to every telephone user. When a user hangs up, her or his bandwidth is assigned to another caller.

Wavelength-Division Multiplexing

Wavelength-division multiplexing (WDM) is designed to use the high-data-rate capability of fiber-optic cable. The optical fiber data rate is higher than the data rate of metallic transmission cable. Using a fiber-optic cable for one single line wastes the available bandwidth. Multiplexing allows us to combine several lines into one. WDM is conceptually the same as FDM, except that the multiplexing and demulti- plexing involve optical signals transmitted through fiber-optic channels. The idea is the same: We are combining different signals of different frequencies. The difference is that the frequencies are very high.

WDM is an analog multiplexing technique to combine optical signals.

Although WDM technology is very complex, the basic idea is very simple. We want to combine multiple light sources into one single light at the multiplexer and do the reverse at the demultiplexer. The combining and splitting of light sources are easily handled by a prism. Recall from basic physics that a prism bends a beam of light based on the angle of incidence and the frequency. Using this technique, a multiplexer can be made to combine several input beams of light, each containing a narrow band of fre- quencies, into one output beam of a wider band of frequencies. A demultiplexer can also be made to reverse the process. One application of WDM is the SONET network in which multiple optical fiber lines are multiplexed and demultiplexed. A new method, called dense WDM (DWDM), can multiplex a very large number of channels by spacing channels very close to one another. It achieves even greater efficiency.

Synchronous Time-Division Multiplexing
Time-division multiplexing (TDM) is a digital process that allows several connections to share the high bandwidth of a link. Instead of sharing a portion of the bandwidth as in FDM, time is shared. Each connection occupies a portion of time in the link.We also need to remember that TDM is, in principle, a digital multiplexing technique. Digital data from different sources are combined into one timeshared link. However, this does not mean that the sources cannot produce analog data; analog data can be sampled, changed to digital data, and then multiplexed by using TDM.

TDM is a digital multiplexing technique for combining several low-rate channels into one high-rate one.

We can divide TDM into two different schemes: synchronous and statistical. We first discuss synchronous TDM and then show how statistical TDM diRers. In synchronous TDM, each input connection has an allotment in the output even if it is not sending data. Time Slots and Frames In synchronous TDM, the data flow of each input connection is divided into units, where each input occupies one input time slot. A unit can be 1 bit, one character, or one block of data. Each input unit becomes one output unit and occupies one output time slot. How- ever, the duration of an output time slot is n times shorter than the duration of an input time slot. If an input time slot is T s, the output time slot is T/n s, where n is the number of connections. In other words, a unit in the output connection has a shorter duration; it travels faster. In synchronous TDM, a round of data units from each input connection is collected into a frame (we will see the reason for this shortly). If we have n connections, a frame is divided into n time slots and one slot is allocated for each unit, one for each input line. If the duration of the input unit is T, the duration of each slot is T/n and the dura- tion of each frame is T (unless a frame carries some other information, as we will see shortly). The data rate of the output link must be n times the data rate of a connection to guarantee the flow of data. the data rate of the link is 3 times the data rate of a connection; likewise, the duration of a unit on a connection is 3 times that of the time slot (duration of a unit on the link). In the figure we represent the data prior to multiplexing as 3 times the size of the data after multiplexing. This is just to convey the idea that each unit is 3 times longer in duration before multiplexing than after.

In synchronous TDM, the data rate of the link is n times faster, and the unit duration is n times shorter.

Time slots are grouped into frames. A frame consists of one complete cycle of time slots, with one slot dedicated to each sending device. In a system with n input lines, each frame has n slots, with each slot allocated to carrying data from a specific input line.

Interleaving

TDM can be visualized as two fast-rotating switches, one on the multiplexing side and the other on the alemultiplexing side. The switches are synchronized and rotate at the same speed, but in opposite directions. On the multiplexing side, as the switch opens in front of a connection, that connection has the opportunity to send a unit onto the path. This process is called interleaving. On the demultiplexing side, as the switch opens in front of a connection, that connection has the opportunity to receive a unit from the path.

Data Rate Management

One problem with TDM is how tO handle a disparity in the input data rates. In all our discussion so far, we assumed that the data rates of all input lines were the same. However, if data rates are not the same, three strategies, or a combination of them, can be used. We call these three strategies multilevel multiplexing, multiple-slot allocation, and pulse stuffing.

Multilevel Multiplexing Multilevel multiplexing is a technique used when the data rate of an input line is a multiple of others. we have two inputs of 20 kbps and three inputs of 40 kbps. The first two input lines can be multiplexed together to provide a data rate equal to the last three. A second level of multi- plexed together to provide a data rate equal to the last three. A second level of multiplexing can create an output of 160 kbps.


Multiple-Slot Allocation Sometimes it is more efficient to allot more than one slot in a frame to a single input line. For example, we might have an input line that has a data rate that is a multiple of another input. the input line with a 50-kbps data rate can be given two slots in the output. We insert a serial-to-parallel converter in the line to make two inputs out of one.

Pulse Stuffing Sometimes the bit rates of sources are not multiple integers of each other. Therefore, neither of the above two techniques can be applied. One solution is to make the highest input data rate the dominant data rate and then add dummy bits to the input lines with lower rates. This will increase their rates. This technique is called pulse stuffing, bit padding, or bit stuffing.

Frame Synchronizing
The implementation of TDM is not as simple as that of FDM. Synchronization between the multiplexer and demultiplexer is a major issue. If the. multiplexer and the demultiplexer are not synchronized, a bit belonging to one channel may be received by the wrong channel. For this reason, one or more synchronization bits are usually added to the beginning of each frame. These bits, called framing bits, follow a pattern, frame to frame, that allows the demultiplexer to synchronize with the incoming stream so that it can separate the time slots accurately. In most cases, this synchronization information consists of 1 bit per frame, alternating between 0 and 1.

Digital Signal Service
Telephone companies implement TDM through a hierarchy of digital signals, called digital signal (DS) service or digital hierarchy.

More Synchronous TDM Applications will be posted after that post.

Digital transmission

A computer network is designed to send information from one point to another. This information needs to be converted to either a digital signal or an analog signal for trans- mission. In this chapter, we discuss the first choice, conversion to digital signals. First, we discuss digital-to-digital conversion tech- niques, methods which convert digital data to digital signals. Second, we discuss analog- to-digital conversion techniques, methods which change an analog signal to a digital
signal. Finally, we discuss transmission modes.

DIGITAL-TO-DIGITAL CONVERSION

The data can be either digital or analog. We also said that signals that represent data can also be digital or analog. In this section, we see how we can represent digital data by using digital signals. The conver- sion involves three techniques: line coding, block coding, and scrambling. Line coding is always needed; block coding and scrambling may or may not be needed.

Line Coding
Line coding is the process of converting digital data to digital signals. We assume that data, in the form of text, numbers, graphical images, audio, or video, are stored in com- puter memory as sequences of bits . Line coding converts a sequence of bits to a digital signal. At the sender, digital data are encoded into a digital signal; at the receiver, the digital data are recreated by decoding the digital signal.

Characteristics
Before discussing different line coding schemes, we address their common characteristics.

Signal Element Versus Data Element Let us distinguish between a data element and a signal element. In data communications, our goal is to send data elements. A data element is the smallest entity that can represent a piece of information: this is the bit. In digital data communications, a signal element carries data elements. A signal element is the shortest unit (timewise) of a digital signal. In other words, data elements are what we need to send; signal elements are what we can send. Data elements are being carried; signal elements are the carriers.

Data Rate Versus Signal Rate
The data rate defines the number of data elements (bits) sent in is. The unit is bits per second (bps). The signal rate is the number of sig- nal elements sent in Is. The unit is the baud. There are several common terminologies used in the literature. The data rate is sometimes called the bit rate; the signal rate is sometimes called the pulse rate, the modulation rate, or the baud rate.
One goal in data communications is to increase the data rate while decreasing the signal rate. Increasing the data rate increases the speed of transmission; decreasing the signal rate decreases the bandwidth requirement. In our vehicle-people analogy, we need to carry more people in fewer vehicles to prevent traffic jams. We have a limited bandwidth in our transportation system.

We now need to consider the relationship between data rate and signal rate (bit rate and baud rate). This relationship, of course, depends on the value of r. It also depends on the data pattern. If we have a data pattern of all 1 s or all Os, the signal rate may be different from a data pattern of alternating Os and 1 s. To derive a formula for the rela- tionship, we need to define three cases: the worst, best, and average. The worst case is when we need the maximum signal rate; the best case is when we need the minimum. In data communications, we are usually interested in the average case. We can formu- late the relationship between data rate and signal rate as

S=c x N x 1/r baud

where N is the data rate (bps); c is the case factor, which varies for each case; S is the
number of signal elements; and r is the previously defined factor. bandwidth with finite values. In other words, the bandwidth is theoretically infinite, but many of the components have such a small amplitude that they can be ignored. The effective bandwidth is finite. From now on, when we talk about the bandwidth of a dig- ital signal, we need to remember that we are talking about this effective bandwidth. We can say that the baud rate, not the bit rate, determines the equired bandwidth for a digital signal. If we use the transportation analogy, the number of vehicles affects the traffic, not the number of people being carried. More changes in the signal mean injecting more frequencies into the signal. (Recall that frequency means change and change means frequency.) The bandwidth reflects the range of frequencies we need. There is a relationship between the baud rate (signal rate) and the bandwidth. Band- width is a complex idea. When we talk about the bandwidth, we normally define a range of frequencies. We need to know where this range is located as well as the values of the lowest and the highest frequencies. In addition, the amplitude (if not the phase) of each component is an important issue. In other words, we need more information about the bandwidth than just its value; we need a diagram of the bandwidth. We will show the bandwidth for most schemes we discuss in the chapter. For the moment, we can say that the bandwidth (range of frequencies) is proportional to the signal rate
(baud rate).

DC Components When the voltage level in a digital signal is constant for a while, the spectrum creates very low frequencies (results of Fourier analysis). These fre- quencies around zero, called DC (direct-current) components, present problems for a system that cannot pass low frequencies or a system that uses electrical coupling (via a transformer). For example, a telephone line cannot pass frequencies below 200 Hz. Also a long-distance link may use one or more transformers to isolate different parts of the line electrically. For these systems, we need a scheme with no DC component.
Self-synchronization To correctly interpret the signals received from the sender, the receiver's bit intervals must correspond exactly to the sender's bit intervals. If the receiver clock is faster or slower, the bit intervals are not matched and the receiver might misinterpret the signals. Figure 4.3 shows a situation in which the receiver has a shorter bit duration. The sender sends 10110001, while the receiver receives 110111000011.

TRANSMISSION MODES

Of primary concern when we are considering the transmission of data from one device to another is the wiring, and of primary concern when we are considering the wiring is the data stream. Do we send 1 bit at a time; or do we group bits into larger groups and, if so, how? The transmission of binary data across a link can be accomplished in either parallel or serial mode. In parallel mode, multiple bits are sent with each clock tick. In serial mode, 1 bit is sent with each clock tick. While there is only one way to send parallel data, there are three subclasses of serial transmission: asynchronous, synchro- nous, and isochronous

Parallel Transmission
Binary data, consisting of ls and Os, may be organized into groups of n bits each. Computers produce and consume data in groups of bits much as we conceive of and use spoken language in the form of words rather than letters. By grouping, we can send data n bits at a time instead of 1. This is called parallel transmission. The mechanism for parallel transmission is a conceptually simple one: Use n wires to send n bits at one time. That way each bit has its own wire, and all n bits of one group can be transmitted with each clock tick from one device to another. The advantage of parallel transmission is speed. All else being equal, parallel transmission can increase the transfer speed by a factor of n over serial transmission.

But there is a significant disadvantage: cost. Parallel transmission requires n communi- cation lines (wires in the example) just to transmit the data stream. Because this is expensive, parallel transmission is usually limited to short distances.

Serial Transmission
In serial transmission one bit follows another, so we need only one communica- tion channel rather than n to transmit data between two communicating devices The advantage of serial over parallel transmission is that with only one communi- cation channel, serial transmission reduces the cost of transmission over parallel by roughly a factor of n. Since communication within devices is parallel, conversion devices are required at the interface between the sender and the line (parallel-to-serial) and between the line and the receiver (serial-to-parallel). Serial transmission occurs in one of three ways: asynchronous, synchronous, and isochronous.

Asynchronous Transmission
Asynchronous transmission is so named because the timing of a signal is unimportant. Instead, information is received and translated by agreed upon patterns. As long as those patterns are followed, the receiving device can retrieve the information without regard to the rhythm in which it is sent. Patterns are based on grouping the bit stream into bytes. Each group, usually 8 bits, is sent along the link as a unit. The sending system handles each group independently, relaying it to the link whenever ready, without regard to a timer. Without synchronization, the receiver cannot use timing to predict when the next group will arrive. To alert the receiver to the arrival of a new group, therefore, an extra bit is added to the beginning of each byte. This bit, usually a 0, is called the start bit. To let the receiver know that the byte is finished, 1 or more additional bits are appended to the end of the byte. These bits, usually 1 s, are called stop bits. By this method, each byte is increased in size to at least 10 bits, of which 8 bits is information and 2 bits or more are signals to the receiver. In addition, the transmission of each byte may then be followed by a gap of varying duration. This gap can be represented either by an idle channel or by a stream of additional stop bits. The start and stop bits and the gap alert the receiver to the beginning and end of each byte and allow it to synchronize with the data stream. This mechanism is called asynchronous because, at the byte level, the sender and receiver do not have to be syn- chronized. But within each byte, the receiver must still be synchronized with the
incoming bit stream. That is, some synchronization is required, but only for the dura- tion of a single byte. The receiving device resynchronizes at the onset of each new byte. When the receiver detects a start bit, it sets a timer and begins counting bits as they come in. After n bits, the receiver looks for a stop bit. As soon as it detects the stop bit, it waits until it detects the next start bit. The addition of stop and start bits and the insertion of gaps into the bit stream make asynchronous transmission slower than forms of transmission that can operate without the addition of control information. But it is cheap and effective, two advan- tages that make it an attractive choice for situations such as low-speed communication. For example, the connection of a keyboard to a computer is a natural application for asynchronous transmission. A user types only one character at a time, types extremely slowly in data processing terms, and leaves unpredictable gaps of time between each character.

Synchronous Transmission
In synchronous transmission, the bit stream is combined into longer "frames," which may contain multiple bytes. Each byte, however, is introduced onto the transmission link without a gap between it and the next one. It is left to the receiver to separate the bit stream into bytes for decoding purposes. In other words, data are transmitted as an unbroken string of 1 s and Os, and the receiver separates that string into the bytes, or characters, it needs to reconstruct the information. The advantage of synchronous transmission is speed. With no extra bits or gaps to
introduce at the sending end and remove at the receiving end, and, by extension, with fewer bits to move across the link, synchronous transmission is faster than asynchro- nous transmission. For this reason, it is more useful for high-speed applications such as the transmission of data from one computer to another. Byte synchronization is accom- plished in the data link layer.
We need to emphasize one point here. Although there is no gap between characters in synchronous serial transmission, there may be uneven gaps between frames.

Isochronous
In real-time audio and video, in which uneven delays between frames are not accept- able, synchronous transmission fails. For example, TV images are broadcast at the rate of 30 images per second; they must be viewed at the same rate. If each image is sent by using one or more flames, there should be no delays between frames. For this type of application, synchronization between characters is not enough; the entire stream of bits must be synchronized. The isochronous transmission guarantees that the data arrive at a fixed rate.

Analog transmission will be posted shortly. (Regards - Utsav Basu)

Cryptography

Cryptography comes from the Greek words for ''secret writing.'' It has a long and colorful history going back thousands of years. In this section we will just sketch some of the highlights, as background information for what follows. For a complete history of cryptography, Kahn's (1995) book is recommended reading. For a comprehensive treatment of the current state-of-the-art in security and cryptographic algorithms, protocols, and applications, see (Kaufman et al., 2002). For a more mathematical approach, see (Stinson, 2002). For a less mathematical approach, see (Burnett and Paine, 2001).

Professionals make a distinction between ciphers and codes. A cipher is a character-for-character or bit-for-bit transformation, without regard to the linguistic structure of the message. In contrast, a code replaces one word with another word or symbol. Codes are not used any more, although they have a glorious history. The most successful code ever devised was used by the U.S. armed forces during World War II in the Pacific. They simply had Navajo Indians talking to each other using specific Navajo words for military terms, for example chay-dagahi-nail-tsaidi (literally: tortoise killer) for antitank weapon. The Navajo language is highly tonal, exceedingly complex, and has no written form. And not a single person in Japan knew anything about it.

In September 1945, the San Diego Union described the code by saying ''For three years, wherever the Marines landed, the Japanese got an earful of strange gurgling noises interspersed with other sounds resembling the call of a Tibetan monk and the sound of a hot water bottle being emptied.'' The Japanese never broke the code and many Navajo code talkers were awarded high military honors for extraordinary service and bravery. The fact that the U.S. broke the Japanese code but the Japanese never broke the Navajo code played a crucial role in the American victories in the Pacific.

Introduction to Cryptography

Historically, four groups of people have used and contributed to the art of cryptography: the military, the diplomatic corps, diarists, and lovers. Of these, the military has had the most important role and has shaped the field over the centuries. Within military organizations, the messages to be encrypted have traditionally been given to poorly-paid, low-level code clerks for encryption and transmission. The sheer volume of messages prevented this work from being done by a few elite specialists.

Until the advent of computers, one of the main constraints on cryptography had been the ability of the code clerk to perform the necessary transformations, often on a battlefield with little equipment. An additional constraint has been the difficulty in switching over quickly from one cryptographic method to another one, since this entails retraining a large number of people. However, the danger of a code clerk being captured by the enemy has made it essential to be able to change the cryptographic method instantly if need be.

The nonsecrecy of the algorithm cannot be emphasized enough. Trying to keep the algorithm secret, known in the trade as security by obscurity, never works. Also, by publicizing the algorithm, the cryptographer gets free consulting from a large number of academic cryptologists eager to break the system so they can publish papers demonstrating how smart they are. If many experts have tried to break the algorithm for 5 years after its publication and no one has succeeded, it is probably pretty solid.

Since the real secrecy is in the key, its length is a major design issue. Consider a simple combination lock. The general principle is that you enter digits in sequence. Everyone knows this, but the key is secret. A key length of two digits means that there are 100 possibilities. A key length of three digits means 1000 possibilities, and a key length of six digits means a million. The longer the key, the higher the work factor the cryptanalyst has to deal with. The work factor for breaking the system by exhaustive search of the key space is exponential in the key length. Secrecy comes from having a strong (but public) algorithm and a long key. To prevent your kid brother from reading your e-mail, 64-bit keys will do. For routine commercial use, at least 128 bits should be used. To keep major governments at bay, keys of at least 256 bits, preferably more, are needed.

From the cryptanalyst's point of view, the cryptanalysis problem has three principal variations. When he has a quantity of ciphertext and no plaintext, he is confronted with the ciphertext-only problem. The cryptograms that appear in the puzzle section of newspapers pose this kind of problem. When the cryptanalyst has some matched ciphertext and plaintext, the problem is called the known plaintext problem. Finally, when the cryptanalyst has the ability to encrypt pieces of plaintext of his own choosing, we have the chosen plaintext problem. Newspaper cryptograms could be broken trivially if the cryptanalyst were allowed to ask such questions as: What is the encryption of ABCDEFGHIJKL?

Novices in the cryptography business often assume that if a cipher can withstand a ciphertext-only attack, it is secure. This assumption is very naive. In many cases the cryptanalyst can make a good guess at parts of the plaintext. For example, the first thing many computers say when you call them up is login: . Equipped with some matched plaintext-ciphertext pairs, the cryptanalyst's job becomes much easier. To achieve security, the cryptographer should be conservative and make sure that the system is unbreakable even if his opponent can encrypt arbitrary amounts of chosen plaintext.

Encryption methods have historically been divided into two categories: substitution ciphers and transposition ciphers. We will now deal with each of these briefly as background information for modern cryptography.

Substitution Ciphers

In a substitution cipher each letter or group of letters is replaced by another letter or group of letters to disguise it. One of the oldest known ciphers is the Caesar cipher, attributed to Julius Caesar. In this method, a becomes D, b becomes E, c becomes F, ... , and z becomes C. For example, attack becomes DWWDFN. In examples, plaintext will be given in lower case letters, and ciphertext in upper case letters.

A slight generalization of the Caesar cipher allows the ciphertext alphabet to be shifted by k letters, instead of always 3. In this case k becomes a key to the general method of circularly shifted alphabets. The Caesar cipher may have fooled Pompey, but it has not fooled anyone since.

The next improvement is to have each of the symbols in the plaintext, say, the 26 letters for simplicity, map onto some other letter. For example,

plaintext: a b c d e f g h i j k l m n o p q r s t u v w x y z

ciphertext: Q W E R T Y U I O P A S D F G H J K L Z X C V B N M

The general system of symbol-for-symbol substitution is called a monoalphabetic substitution, with the key being the 26-letter string corresponding to the full alphabet. For the key above, the plaintext attack would be transformed into the ciphertext QZZQEA.

At first glance this might appear to be a safe system because although the cryptanalyst knows the general system (letter-for-letter substitution), he does not know which of the 26! 4 x 10²⁶ possible keys is in use. In contrast with the Caesar cipher, trying all of them is not a promising approach. Even at 1 nsec per solution, a computer would take 10¹⁰ years to try all the keys.

Nevertheless, given a surprisingly small amount of ciphertext, the cipher can be broken easily. The basic attack takes advantage of the statistical properties of natural languages. In English, for example, e is the most common letter, followed by t, o, a, n, i, etc. The most common two-letter combinations, or digrams, are th, in, er, re, and an. The most common three-letter combinations, or trigrams, are the, ing, and, and ion.

A cryptanalyst trying to break a monoalphabetic cipher would start out by counting the relative frequencies of all letters in the ciphertext. Then he might tentatively assign the most common one to e and the next most common one to t. He would then look at trigrams to find a common one of the form tXe, which strongly suggests that X is h. Similarly, if the pattern thYt occurs frequently, the Y probably stands for a. With this information, he can look for a frequently occurring trigram of the form aZW, which is most likely and. By making guesses at common letters, digrams, and trigrams and knowing about likely patterns of vowels and consonants, the cryptanalyst builds up a tentative plaintext, letter by letter.

Another approach is to guess a probable word or phrase. For example, consider the following ciphertext from an accounting firm (blocked into groups of five characters):

CTBMN BYCTC BTJDS QXBNS GSTJC BTSWX CTQTZ CQVUJ
QJSGS TJQZZ MNQJS VLNSX VSZJU JDSTS JQUUS JUBXJ
DSKSU JSNTK BGAQJ ZBGYQ TLCTZ BNYBN QJSW 

A likely word in a message from an accounting firm is financial. Using our knowledge that financial has a repeated letter (i), with four other letters between their occurrences, we look for repeated letters in the ciphertext at this spacing. We find 12 hits, at positions 6, 15, 27, 31, 42, 48, 56, 66, 70, 71, 76, and 82. However, only two of these, 31 and 42, have the next letter (corresponding to n in the plaintext) repeated in the proper place. Of these two, only 31 also has the a correctly positioned, so we know that financial begins at position 30. From this point on, deducing the key is easy by using the frequency statistics for English text.

Quantum Cryptography

Interestingly, there may be a solution to the problem of how to transmit the one-time pad over the network, and it comes from a very unlikely source: quantum mechanics. This area is still experimental, but initial tests are promising. If it can be perfected and be made efficient, virtually all cryptography will eventually be done using one-time pads since they are provably secure. Below we will briefly explain how this method, quantum cryptography, works. In particular, we will describe a protocol called BB84 after its authors and publication year (Bennet and Brassard, 1984).

A user, Alice, wants to establish a one-time pad with a second user, Bob. Alice and Bob are called principals, the main characters in our story. For example, Bob is a banker with whom Alice would like to do business. The names ''Alice'' and ''Bob'' have been used for the principals in virtually every paper and book on cryptography in the past decade. Cryptographers love tradition. If we were to use ''Andy'' and ''Barbara'' as the principals, no one would believe anything in this chapter. So be it.

If Alice and Bob could establish a one-time pad, they could use it to communicate securely. The question is: How can they establish it without previously exchanging DVDs? We can assume that Alice and Bob are at opposite ends of an optical fiber over which they can send and receive light pulses. However, an intrepid intruder, Trudy, can cut the fiber to splice in an active tap. Trudy can read all the bits in both directions. She can also send false messages in both directions. The situation might seem hopeless for Alice and Bob, but quantum cryptography can shed some new light on the subject.

Quantum cryptography is based on the fact that light comes in little packets called photons, which have some peculiar properties. Furthermore, light can be polarized by being passed through a polarizing filter, a fact well known to both sunglasses wearers and photographers. If a beam of light (i.e., a stream of photons) is passed through a polarizing filter, all the photons emerging from it will be polarized in the direction of the filter's axis (e.g., vertical). If the beam is now passed through a second polarizing filter, the intensity of the light emerging from the second filter is proportional to the square of the cosine of the angle between the axes. If the two axes are perpendicular, no photons get through. The absolute orientation of the two filters does not matter; only the angle between their axes counts.

To generate a one-time pad, Alice needs two sets of polarizing filters. Set one consists of a vertical filter and a horizontal filter. This choice is called a rectilinear basis. A basis (plural: bases) is just a coordinate system. The second set of filters is the same, except rotated 45 degrees, so one filter runs from the lower left to the upper right and the other filter runs from the upper left to the lower right. This choice is called a diagonal basis. Thus, Alice has two bases, which she can rapidly insert into her beam at will. In reality, Alice does not have four separate filters, but a crystal whose polarization can be switched electrically to any of the four allowed directions at great speed. Bob has the same equipment as Alice. The fact that Alice and Bob each have two bases available is essential to quantum cryptography.

For each basis, Alice now assigns one direction as 0 and the other as 1. In the example presented below, we assume she chooses vertical to be 0 and horizontal to be 1. Independently, she also chooses lower left to upper right as 0 and upper left to lower right as 1. She sends these choices to Bob as plaintext.

Now Alice picks a one-time pad, for example based on a random number generator (a complex subject all by itself). She transfers it bit by bit to Bob, choosing one of her two bases at random for each bit. To send a bit, her photon gun emits one photon polarized appropriately for the basis she is using for that bit. For example, she might choose bases of diagonal, rectilinear, rectilinear, diagonal, rectilinear, etc. To send her one-time pad of 1001110010100110 with these bases.

Two Fundamental Cryptographic Principles

Although we will study many different cryptographic systems in the pages ahead, two principles underlying all of them are important to understand.

Redundancy

The first principle is that all encrypted messages must contain some redundancy, that is, information not needed to understand the message. An example may make it clear why this is needed. Consider a mail-order company, The Couch Potato (TCP), with 60,000 products. Thinking they are being very efficient, TCP's programmers decide that ordering messages should consist of a 16-byte customer name followed by a 3-byte data field (1 byte for the quantity and 2 bytes for the product number). The last 3 bytes are to be encrypted using a very long key known only by the customer and TCP.

At first this might seem secure, and in a sense it is because passive intruders cannot decrypt the messages. Unfortunately, it also has a fatal flaw that renders it useless. Suppose that a recently-fired employee wants to punish TCP for firing her. Just before leaving, she takes the customer list with her. She works through the night writing a program to generate fictitious orders using real customer names. Since she does not have the list of keys, she just puts random numbers in the last 3 bytes, and sends hundreds of orders off to TCP.

When these messages arrive, TCP's computer uses the customer's name to locate the key and decrypt the message. Unfortunately for TCP, almost every 3-byte message is valid, so the computer begins printing out shipping instructions. While it might seem odd for a customer to order 837 sets of children's swings or 540 sandboxes, for all the computer knows, the customer might be planning to open a chain of franchised playgrounds. In this way an active intruder (the ex-employee) can cause a massive amount of trouble, even though she cannot understand the messages her computer is generating.

This problem can be solved by the addition of redundancy to all messages. For example, if order messages are extended to 12 bytes, the first 9 of which must be zeros, then this attack no longer works because the ex-employee can no longer generate a large stream of valid messages. The moral of the story is that all messages must contain considerable redundancy so that active intruders cannot send random junk and have it be interpreted as a valid message.

However, adding redundancy also makes it easier for cryptanalysts to break messages. Suppose that the mail order business is highly competitive, and The Couch Potato's main competitor, The Sofa Tuber, would dearly love to know how many sandboxes TCP is selling. Consequently, they have tapped TCP's telephone line. In the original scheme with 3-byte messages, cryptanalysis was nearly impossible, because after guessing a key, the cryptanalyst had no way of telling whether the guess was right. After all, almost every message is technically legal. With the new 12-byte scheme, it is easy for the cryptanalyst to tell a valid message from an invalid one. Thus, we have

Cryptographic principle 1: Messages must contain some redundancy

In other words, upon decrypting a message, the recipient must be able to tell whether it is valid by simply inspecting it and perhaps performing a simple computation. This redundancy is needed to prevent active intruders from sending garbage and tricking the receiver into decrypting the garbage and acting on the ''plaintext.'' However, this same redundancy makes it much easier for passive intruders to break the system, so there is some tension here. Furthermore, the redundancy should never be in the form of n zeros at the start or end of a message, since running such messages through some cryptographic algorithms gives more predictable results, making the cryptanalysts' job easier. A CRC polynomial is much better than a run of 0s since the receiver can easily verify it, but it generates more work for the cryptanalyst. Even better is to use a cryptographic hash, a concept we will explore later.

Getting back to quantum cryptography for a moment, we can also see how redundancy plays a role there. Due to Trudy's interception of the photons, some bits in Bob's one-time pad will be wrong. Bob needs some redundancy in the incoming messages to determine that errors are present. One very crude form of redundancy is repeating the message two times. If the two copies are not identical, Bob knows that either the fiber is very noisy or someone is tampering with the transmission. Of course, sending everything twice is overkill; a Hamming or Reed-Solomon code is a more efficient way to do error detection and correction. But it should be clear that some redundancy is needed to distinguish a valid message from an invalid message, especially in the face of an active intruder.

Freshness

The second cryptographic principle is that some measures must be taken to ensure that each message received can be verified as being fresh, that is, sent very recently. This measure is needed to prevent active intruders from playing back old messages. If no such measures were taken, our ex-employee could tap TCP's phone line and just keep repeating previously sent valid messages. Restating this idea we get.

Cryptographic principle 2: Some method is needed to foil replay attacks

One such measure is including in every message a timestamp valid only for, say, 10 seconds. The receiver can then just keep messages around for 10 seconds, to compare newly arrived messages to previous ones to filter out duplicates. Messages older than 10 seconds can be thrown out, since any replays sent more than 10 seconds later will be rejected as too old.

Subnet Masks

Introduction to Subnet Masks

Subnet masks are one of the most interesting aspects of TCP/IP. Subnet masks point out to IP which bits of the 32-bit IP address refer to the network. A good network administrator understands how to determine and use subnet masks.


What Is a Subnet Mask?

A subnet mask is a number that looks like an IP address. It shows TCP/IP how many bits are used for the network portion of the IP address by covering up, or “masking,” the IP address’s network portion. As you learned in Chapter 6, an IP address is made up of two parts: the network portion and the host portion. For every outgoing packet, IP has to determine whether the destination host is on the same local network or on a remote network . If the destination is local, then IP uses an ARP broadcast to find out the hardware address of the destination host. If the destination host is not on the local network, then ARP broadcastsa request for the hardware address of the router. Therefore, IP sends packets that are bound for a remote network directly to the router, which is also known as the default gateway. The router then sends the packet to the next network on its journey to the correct destination network.Just as the telephone system uses an area code to determine whether a number is local or long distance, TCP/IP uses the subnet mask to determine whether the destination of a packet is a host on the local network or a host on a remote network. In the same way that every U.S. telephone number must have an area code, every IP address must have a subnet mask. If, for example, your telephone number is (619) 555-1212, and you call someone whose telephone number is (619) 345-1111, it is a local call. You know that because you can look at the numbers between the parentheses and see that they have the same value. If, on the other hand, your number is (619) 555-1212, and you call someone whose number is (213) 888-8146, it’s a long distance call. You know that because the numbers inside of the parentheses are different. You can think of the subnet mask as the area code in the parentheses of a telephone number. Just as an area code determines a phone call’s destination, a subnet mask tells IP how many bits to look at when determining if the destination IP address is local or remote.The following graphic shows Harry calling Amber. Since Amber has a different area code, the phone call will have to go through the router. When Harry calls Sally, however, it is a local call and does not need to go through the router. When determining if the packet is bound for the local network or a remote network, IP compares the network portion of the sender’s IP address with the same number of bits from the destination’s IP address. If the bit values are exactly the same, the packet’s destination is determined to be local. If there are
any differences in the bit values, the packet’s destination is determined to be remote. To know how many bits to compare, IP evaluates the subnet mask of the sending host. In the subnet mask, there is a series of 1s, and then the rest of the bits are set to 0. When IP evaluates the subnet mask, it is looking specifically for the answer to the question, “How many bits are set to 1?” Once IP determines how many bits are set to 1, it knows how many bits of the source host’s IP address and the destination host’s IP address will be compared.You can think of the number of bits that are set to 1 in the subnet mask as the number of digits inside the parentheses in a telephone number—if that number could change (in other words, if it’s variable). If, for example, a telephone number has 10 digits, imagine if the parentheses include 4, 5, or 6 digits. You would
then evaluate the number to be local or long distance based on the digits that are in the arentheses. If there are 8 bits set to 1 in the subnet mask, IP will compare the first 8 bits of the host with the first 8 bits of the destination. If there are 16 bits in the subnet mask that are set to 1, IP will compare the first 16 bits of host and destination. A subnet mask is a required element of every IP address. When you want to type in the IP address for a host, the only two required elements are the IP address itself and the subnet mask. Likewise, when you want to call someone, it is required that you know the correct area code for the phone number. You then
compare the first three characters of your phone number (your area code) with the first three characters of their phone number (their area code). If the area codes are the same, you don’t need to dial the area code, nor do you have to pay for a long distance call, because it is a local call. If the area code is not the same, however, you’ll have to dial their area code so that the telephone system can route your call to their city. You’ll see over the next several pages that IP looks at everything in binary. Subnet masks and routing will become clearer if you think about the IP addresses and subnet masks in binary, so begin now to think of IP addresses and subnet
masks as 32 bits. When thinking in binary, do not pay attention to the periods
that we use in the decimal representation. IP does not pay attention to the periods;
neither should we. Just consider the addresses as 32 1s and 0s.

Network and Host

subnet goggles
A fictional set of goggles that IP wears
when looking at an IP address to determine
whether an address is local or
remote. The goggles “light up” the network
and subnet bits with 1s as the bit
values in the subnet mask.
Applying a subnet mask is like looking through a set of “
subnet goggles
.” Imagine
wearing a set of goggles as you look at an IP address; you see all 32 bits, each
in its own slot. When you ask the question, “How many bits are used for the network
portion of this IP address?” the subnet mask lights up the slots that are in
the network portion of the address.
Through subnet goggles, 255.0.0.0 looks like this:
NNNN NNNN.
HHHH HHHH.HHHH HHHH.HHHH HHHH
The goggles light up the first 8 bits as the network portion (
N
), and the
remaining 24 bits are used for the host portion (H).
Through subnet goggles, 255.255.0.0 looks like this:
NNNN NNNN.NNNN NNNN.
HHHH HHHH.HHHH HHHH
The goggles light up the first 16 bits as the network portion.
The subnet mask simply provides a means to light up the correct slots so that
IP can figure out the number of bits used for the network portion of the address.
After IP figures this out, it can compare the address to that of another host to
determine whether that host is local or remote. Using our telephone number and
area code example, we can say that the subnet goggles are illuminating the area

Identifying a Local or Remote Network

With every packet that is sent across a network, the big question is: Is the destination
address local or remote? The destination is local if the network portion of the source’s IP address is the same as that of the destination’s IP address. If any bits of the network portions differ from each other, then the destination
is remote. This is similar to figuring out whether someone lives on the same street as you do. If you look at the person’s street name and it is the same as yours, the person lives on the same street as you do. If any part of the street name is different, the person is remote to your street. But, as stated earlier, before IP can figure out whether the destination address is remote, IP has to determine how many bits are in the network portion of the source IP address. IP uses the subnet mask to determine which bits of the IP address represent the network portion of the address.The subnet mask is 32 bits long, but you use dotted decimal notation to represent
it, just as you do with an IP address. A subnet mask, in binary, is made up of several contiguous 1s, which represent the network portion of the address, and then the rest of the bits are 0s. When determining how many of the 32 bits are in the network portion of an IP address, IP looks at the subnet mask for the contiguous 1s. When you look at a subnet mask in binary, imagine that the 1s represent the beginning and end of an area code. The number of bits set to 1 in the subnet mask is the number of bits that will be compared to determine if the destination is local or remote. This is similar to evaluating two telephone numbers by comparing the values that are inside the parentheses. The 1s in the subnet mask will act like the number of digits within the parentheses in an area code; these are the only values that are compared to determine if the destination is local or remote. When someone gives you their telephone number, you can tell if it is a long distance number just by looking at the digits in the parentheses. Likewise, the subnet mask’s only purpose is to determine how many bits are used to identify if the destination host of every packet is local or remote. For example, if the first 16 bits are set to 1, then IP compares the first 16 bits of the source IP address with the first 16 bits of the destination IP address. If these 16 bits are exactly the same, the destination host is local; if any of the bits are different,the destination host is remote. If the first 24 bits are set to 1, then IP compares the first 24 bits of the source IP address with the first 24 bits of the destination IP address. If these 24 bits are exactly the same, the destination host is local; if any of the bits are different, the destination host is remote. It is called a subnet “mask” for a good reason: it indicates or “masks” the network bits. Think of it as a shadow covering up some of the bits.

Standard Subnet Masks

In Chapter 6, you looked at the five classes of IP addresses. For each class of address, there is a standard, or default, subnet mask. Each is discussed in the following sections.

Class A Addresses

The standard subnet mask for a Class A address is 255.0.0.0. This tells IP that the first 8 bits are used for the network portion of the IP address, and the remaining 24 bits are used for the host portion. IP looks at the 32 bits and uses the subnet mask to mask out the network portion of the address: NNNN NNNN.HHHH HHHH.HHHH HHHH.HHHH HHHH

Because 24 bits are left for the host portion of the address, there are almost 17 million unique host IP addresses for each Class A network address.

Class B Addresses

A Class B address has a standard subnet mask of 255.255.0.0. This mask tells IP that the first 16 bits are used for the network portion of the address, and the remaining 16 bits are used for the host portion: NNNN NNNN.NNNN NNNN.HHHH HHHH.HHHH HHHH The 16 bits that are used for the host portion of the address can uniquely address more than 16,000 hosts on each Class B network.

Class C Addresses

A Class C address has a standard subnet mask of 255.255.255.0, which masks out the first 24 bits as the network portion and leaves the remaining 8 bits for the host portion:
NNNN NNNN.NNNN NNNN.NNNN NNNN. HHHH HHHH The 8 bits used for the host portion can uniquely address 254 hosts on each of the Class C networks.

In Summary

Class Subnet Masks(decimal) Standard Masks (Binary)


A 255.0.0.0 1111 1111.0000 0000.0000 0000.0000 0000

B 255.255.0.0 1111 1111.1111 1111.0000 0000.0000 0000

C 255.255.255.0 1111 1111.1111 1111.1111 1111.0000 0000

You can remember the standard masks this way:
1 octet = Class A (1st letter in the alphabet)
2 octets = Class B (2nd letter in the alphabet)
3 octets = Class C (3rd letter in the alphabet)
In most cases, however, using the standard subnet mask is not the optimal
solution for designing a TCP/IP addressing plan. Most implementations use a
variation of the standard subnet mask called a custom subnet mask, which is
explained in Chapter 9, “Using Custom Subnet Masks.”
The following screen capture shows a custom subnet mask being used.
Because the IP address has “10” in the first octet, this is a Class A address, and
the standard subnet mask is 255.0.0.0. However, the administrator has defined
a custom subnet mask of 255.255.255.240, which enables him to create more
networks with fewer hosts on each network.


Custom Subnet Mask will be posted later. (Regards - Utsav)

FTP/HTTP

An application used to transfer files from
one host to another and to store the files
on the requesting host.

File Transfer Protocol (FTP)

is the protocol that defines how a file can be transferred
from one host to another. For a file to be transferred from one host to
another, the FTP on the initiating host creates the request for a file, and FTP on
the FTP server processes the requests for a file. A programmer who would like to
learn all of the idiosyncrasies of FTP should read RFC 959.Two hosts are involved in an FTP session. One host requests a file, and the
other host has a copy of the file and transfers a copy to the requesting host. Files
can be transferred in either a text or binary format.
The host that is requesting the service is called a client, and the host that provides
the service is called a server. These two hosts establish a client/server relationship,
which is simply one host making requests of another.The requesting host uses an application to request the file. The application
may be a word processor, an
FTP command-line utility
, or an
FTP command
interpreter
. The FTP command-line utility enables a host to connect to an
FTP
server
without using a fancy interface by having the user simply enter FTP commands
at the command line. FTP connects to the FTP server, and the user is requested to log in.The user must supply a username and a password. In the screen capture below, a connection was made to an FTP server at ftp.microsoft.com . In this
FTP session, the user logged in with the account name ftp and no password
Most FTP sites will allow an anonymous user to log in with no password.
When prompted for a username, you can type anonymous . However, because
anonymous is so difficult to spell, you might want to log in by typing
ftpas your anonymous username. Although no password is required, using your e-mail
address as a password is considered “good FTP etiquette” when using the anonymous
account.How FTP Works
The first packet that is sent from the requesting host to the FTP server is a
TCP/IP packet requesting to set up a connection. In this example, the packet was
sent to TCP port 21 on the FTP server. The requesting host chose a non-wellknown
port to listen for the reply. In this case, the requesting host chose 1177 as
the return port and will be listening for a response sent to that port number. In
the next screen capture, the first TCP/IP packet is shown as the requesting host
makes a request from the source port of 1177 to the destination port of 21.
The FTP application was listening at port 21. Upon receiving the request, the
application sent back an
FTP/TCP/IP
packet to set up the connection and ask
that the client send back the username to log in. In this return packet, the FTP
server is the source, and the FTP client is the destination. The FTP server sends
this reply to port 1177, where the FTP client said it would be listening. In the FTP
header that the FTP server built, the FTP server is passing FTP information to the
FTP client. Notice in the screen capture that the source port labeled src:is set to port
1177. Harry, the requesting host, has decided that the FTP server should send
data back to this port. The following screen capture shows the next packet in the sequence, where the FTP server replies to the FTP client.
This dialog continues as the FTP server responds to the FTP client’s requests.
The client will be able to see a list of files that are available and request either one
or more server files be transferred.The command-line FTP client application requires that the user know the FTP
commands and how to use them. Another way that the user can connect to an
FTP server—without knowing how to use the commands—is to use an FTP command
interpreter. Several of these interpreter applications are available on the
Internet; some are shareware, and some can be downloaded and used for a trial
period. An example of one that can be downloaded for a 30-day trial before you
buy it is
CuteFTP
. CuteFTP has an easy-to-learn and easy-to-use interface. This
client application interprets the user’s clicks, translates them to FTP commands,
and passes those commands to the FTP server. Another example of an FTP command
interpreter available on the Internet is FTP Voyager.

For FTP to work, the server must be running an FTP server application, and the client must be using an FTP client application.

Hypertext Transfer Protocol (HTTP)

it is a set of rules for exchanging files on the Internet. This is the protocol that your Web browser uses when surfing the Internet.

Unlike FTP, HTTP is designed so that very little user intervention is required. HTTP
transfers preformatted files that are displayed in their browser instead of just saved
to disk. The HTTP application runs on a Web server and listens for requests, and
then responds by sending files back to the requestor. A Web server is a server that has

the HTTP service application running on it. HTTP listens at a TCP port, usually port
80 for requests, and then transfers the requested file back to the requestor. The requesting host displays the file in a Web browser application. The client makes the HTTP request by issuing a command to their Web browser. The command is initiated by typing a Uniform Resource Locator (URL) , (such as www.ep6network.blogspot.com) in the address line of the Web browser or by clicking a hyperlink on a page that is being displayed by the Web browser. The Web browser formats the client’s request into an HTTP/TCP/IP request packet with a destination port of 80.At the Web server, the HTTP application is listening at port 80 for any requests.
After the packet is received, the appropriate file is retrieved and packaged for delivery
to the client. The packets leave the Web server, and upon arrival at the client,
the Web browser decodes the Hypertext Markup Language (HTML) file and displays
it onscreen with the proper formatting.So, let’s look at what is really happening when you connect to a Web site:
1.You open your Web browser and type in the URL
www.ep6network.blogspot.com.

2.Your Web browser creates the TCP/IP packet and sends it to a Web server
somewhere on the Internet. In other words, little ol’ you makes a request
of a big Web server to set up a connection.
3.The Web server hears your request at port 80 and sends back a packet to
you that says, “Okay, I’ll set up a connection with you.”

4.Now that you have a connection with the Web server, you request that the
Web server send you its default page.

5.The Web server receives your response and gets the file that you
requested. The file is put into one or more packets, depending on how
big the file is, and it is sent to you.

6.Your Web browser receives the packets and sends back an acknowledgment
that they were received. If the Web server does not get an acknowledgment
from you, the packet is re-sent.

7.Your Web browser displays the information that you requested on your
screen as the packets are received.

Ports and Firewalls
Every packet that travels on the network contains several pieces of information
that is used to ensure that it arrives at its destination. Part of the information in
the header provides the destination hardware (MAC) address, another part provides
the destination IP address, and another part provides the destination port.
As a packet is gaining admission to a network and before a packet reaches its destination,
a firewall can be used to protect the destination host from a malicious
packet. A firewall will disallow packets with a destination address and port that
are not permitted.
Requesting a Service in the TCP/IP Stack
Imagine the TCP/IP stack as an extremely tall office building with two towers.
The two towers share the same lobby and each tower has 65,536 floors. Because
they are part of the same building, these two towers have the same physical or
street address. The street address in this analogy is the hardware (MAC) address.
Suppose you want to make a request of a company located in this building. To
do so, you need to know the address of the building and the floor on which the
company does business. Using the street address helps you to arrive at the correct
building. Once in the lobby, you make your way to the elevators so that you can
travel up to the company’s floor. As you arrive at the bank of elevators, you need
to know which set of elevators to use. One tower is the TCP tower and the other
is the UDP tower. For this example, you need to access TCP port 80 to find the
company that you’ve come here to do business with. You get on the TCP elevator
and go to the 80th floor. As the elevator door opens, a concierge meets you to
handle your service requests. You make your service request, and the concierge
begins to process your request.
As discussed earlier in this chapter, TCP port 80 is the port where an HTTP service
listens for requests. As the elevator door opens at the 80th floor, it is an HTTP
service that waits to process your request. If you access the 21st floor, an FTP service
will be waiting to process your request. When programmers write a service,
they determine which port they want the service to be listening at. When a client
sends a packet to the server that is hosting the service, the client must make a
request of the correct port. If the service is listening at port 80, and a client makes
a request at port 81, the request will not be processed because there is no service
listening at port 81.Imagine our towers and all of the empty floors. A server does not run 65,536
services, and definitely not in both the TCP and UDP buildings. With all these
empty floors, we don’t want visitors to get off the elevators at floors that are not
occupied, or not running a service. If there are bad guys out there trying to infiltrate
our building, they would probably try to enter through a floor where no service
and no security measures have been set up. On the floors where we have a
service running, the service has a level of built-in security, but on the floors with
no service running, an intruder could access the floor.

The Firewall is Protecting the LAN

On a Local Area Network (LAN), every host is like the TCP/IP building that was just
described. To protect the LAN from intruders, most administrators implement a
firewall
. A firewall is a combination of hardware and software that is installed at the
edge of the LAN. The firewall works like a military checkpoint on the edge of a city.
An administrator puts a firewall at every entrance to the LAN. Every packet that tries
to make its way onto the LAN must pass through the checkpoint and be inspected.
Most firewalls are set up to stop all traffic initially, and the administrator configures
rules to allow certain traffic onto the LAN. The administrator may also configure the
firewall to deny certain traffic from leaving the LAN.
Picture an island with a city full of TCP/IP buildings, and the only way to get on
or off the island is to cross a bridge. This connection to the island—the bridge—is
where a checkpoint belongs. If there are other ways to access the island, a checkpoint
must be at each path. On a network, a firewall must be placed at every
entrance or exit to the Internet. At the checkpoints, each packet is examined, and
based on the rules established by the administrator, the packet is either allowed or
denied access.
These rules are set up based on IP addresses, DNS names, protocols, and ports.
The administrator creates rules based on the services that the administrator wants
to make available to other networks. For example, without an FTP server on the
LAN, the administrator will deny any inbound packet that is trying to connect to
port 21. If the LAN has an FTP server, and the administrator would like someone
outside of the LAN to have access to the FTP server, the administrator can create
a rule that allows a packet addressed to the server and port 21 to enter the LAN.
Without a firewall, your TCP/IP buildings are not secure. All packets are allowed
to enter the island and access the destination host. When a packet is allowed access
to the LAN, the packet can arrive at the destination host and travel up to the 21st
floor. Even though no service is running on the 21st floor, the elevator door will
open, and the packet will have access to the host.
Every network must be secured by a firewall. An administrator can also put
a firewall right in front of a server or running on the server so that the server will
be double protected. With several firewall products available, administrators
need to evaluate and implement the best solution for their networks. This evaluation
includes taking a good look at the how sensitive or critical the data is and
how much money is in the budget for the firewall solution.

Some Web sites on the Internet will test your machine’s vulnerability to TCP/IP
attacks. From your favorite search engine, search for “firewall vulnerability test,” and
you should find a couple of free, online vulnerability testing sites.

Broadband

Always-on Access

Before the late 1990s, people connected remotely to their offices or the Internet using dial up connections. An always-on remote network connection was not possible for a reasonable price. To connect to the corporate network, the user ran a program that dialed a phone number. Unless the user had a second phone line, being online prohibited incoming or outgoing phone calls. The user entered a user ID and password to gain access to the system. The fastest speed available over phone lines was 56 kbps, which was fine until the web became popular in the 1990s. Downloading large pictures, documents, applications, and audio files took what seemed like forever. Then, along came broadband. Broadband networking offered a reasonable high-speed alternative to traditional dialup networking. Using existing service connections to houses (such as phone wiring, cable TV coaxial cable, or even satellite), service providers offered Internet services at many times the speed of dialup. Downloading large files became palatable with broadband. Broadband technologies allow service providers to offer always-on connectivity similar to what people use in a corporate network. Computers on the broadband network always have access to the network; there is no intermediate dialup step. Sit down, load the browser, and off you go. High-speed Internet access to homes offers new levels of productivity and entertainment not possible before the commercialization of the Internet and the web. Aside from apparent uses such as online shopping and video streaming, corporations can accommodate road warriors and work-from-home folks in a way not previously possible. Using encryption technologies, an employee with a laptop computer can securely access her corporate network from any Internet access point in the world. Additionally, employees can attach IP phones, allowing them to work on their computers and make calls from their office-phone extensions as if they were sitting at their desks.

Broadband Technology Evolution

Integrated Services Digital Network (ISDN) was the first commercially viable broadband option available. Using existing phone lines, home users commonly subscribed to a Basic Rate Interface (BRI), which had a throughput maximum of 128 kbps. ISDN had some significant adoption in Europe, but in the U.S., ISDN was eclipsed by more cost-effective broadband technologies before it had a chance to become commonplace. Cable modem and digital subscriber line (DSL) services became the premier broadband technologies. Although other broadband technologies existed, the primary determination of a technology’s viability was access to “last mile” wiring to houses. Anything that required new wiring probably wouldn’t make it. Other technologies that take advantage of other media exist, such as satellite television dishes, but they did not become widely adopted. For those requiring even higher throughput, some providers now offer fiber links to homes as a premier service. This is already popular throughout major cities in Asia and is being installed in several cities in the U.S. as well.

Cable Modem

Cable modems provide high-speed data communication using existing cable television coaxial cabling. Current implementations of cable-modem technologies offer speeds as fast as Ethernet (greater than 10 Mbps). This means that a file that takes 2 minutes to transfer over ISDN takes 2 seconds over a cable modem. Cable modem can provide higher speeds than traditional leased lines, with lower cost and easier installation. Because a cable-modem connection is permanently established, it cannot dial multiple locations directly. As a result, cable-modem access must be to the Internet. This restriction means that employees can connect to their company’s network only if the company provides access through the Internet. Usually, this is done through a secure VPN connection.

DSL

Like cable modems, DSL provides high-speed Internet access for reasonable cost using existing cabling to houses and businesses. DSL carves off a portion of the telephone line to use for data transmission without interfering with existing phone service. Because of the multiple flavors of DSL services, DSL is generically referred to as xDSL. The two popular forms of xDSL service currently available are Asymmetric DSL (ADSL) and Symmetric DSL (SDSL). ADSL provides faster download speeds because traffic toward the user is given more bandwidth than traffic from the user. SDSL assigns equal bandwidth in both directions. ADSL is most often used for residential service, and SDSL is most often used in commercial settings, because of their different usage models.

Which One Is Better?

Both DSL and cable modems provide high-speed Internet access at a relatively low cost. Both provide always-on connectivity. Both have technical advantages and disadvantages. Either technology makes a good to-the-home or small office solution for Internet connectivity. Because both technologies are always on, a firewall must protect the local network from Internet-based attacks. Some practical issues affect how widespread the technologies become. Virtually all businesses and homes have telephone lines, which means that DSL is possible, but fewer homes and businesses have cable TV connections. In general, neither is “better,” and both types of service offer very high-speed connectivity for relatively low cost.

Digital Subscriber Line (DSL)

DSL uses the existing phone wires connected to virtually every home in most countries. The twisted-pair wires that provide phone service are ideal, because the available frequency ranges on the wires far exceed those required to carry a voice conversation. Human speech occupies frequencies of roughly 4000 hertz (4 kHz) or less. The copper wires that provide phone service can carry in the range of 1 to 2 million hertz (1 to 2 MHz). DSL provides more downstream data (from the Internet to you) than upstream data (from you to the Internet) based on user profiles, but this can be changed for businesses or those running web servers.

DSL Equipment

DSL requires some specialized equipment to ensure that the voice and data are kept separate and are routed to the right place. • Low-pass filters (LPF) are placed on all phone jacks not used by a computer to prevent interference from high-frequency data signals. DSL modems are the interface from the phone line to the computer. • DSL access multiplexers (DSLAM) aggregate hundreds of signals from homes and are the access point to the Internet.

Limitations and Advantages

DSL signals are distance-sensitive, which means that the available throughput decreases the farther your house is from the service provider. The maximum distance is about 18,000 feet. DSL signals cannot be amplified, nor can they be converted from one medium to another between the DSL modem and the DSLAM. (For example, opticalfiber extensions are not possible.) Typically the DSL company performs a line test to ensure that the service can be supported at a particular residence. The good news for DSL is that throughput is unaffected by the number of users so long as the phone company continues to add DSLAMs to support
new users.

Cable

Cable uses the same basic principle as DSL in that the bandwidth needed to accomplish the primary function is only a fraction of the available bandwidth on the wire or, in this case, cable. Cable is a slightly different concept in how it divides the available frequencies. The cable spectrum is already divided up into several hundred 6-MHz blocks to account for the various cable channels. Your cable-ready TV simply tunes its receiver to the frequency that corresponds
to the channel you have chosen. To add Internet capabilities, each user is assigned one or more blocks for downstream data (each 6-MHz block is good for about 30 Mbps of data). For the upstream piece, the lower end of the spectrum is divided into 2-MHz blocks, because most people download more information than they upload. Each subscriber is assigned one or more 2-MHz blocks.

Wan network

Moving Traffic Across the Street and the World

How does a company connect the network in its New York office to the network
in its Los Angeles office? It doesn’t make sense to run a private cable
across the U.S. Instead, the company subscribes to wide-area services. A widearea
network (WAN) is a network that covers a broad geographic area and
often uses transmission facilities provided by service providers. WAN functionality
occurs at Layers 1 to 3 in the Open Systems Interconnection (OSI) reference
model.
The bicoastal company just mentioned would contact its service provider to
purchase WAN connectivity between the offices. WAN services are leased from
service providers that charge either a monthly flat fee or fees based on the
amount of data transmitted. The more bandwidth required for a WAN circuit,
the greater the usage charges are. Service providers can use a single national
network to provide WAN services for many different corporate customers. In
this way it is not necessary for each company to physically connect every office
to every other office. Imagine the cross-country cables involved in connecting
just one large company, let alone thousands.

WAN Services

Three types of transport are used with WANs:
• Point-to-point: Also known as leased line, a point-to-point connection is a
pre-established link from one site, across a service provider’s network, to a
remote site. The carrier establishes the point-to-point link for the customer’s
private use.
• Circuit switching: A service provider establishes a dedicated physical circuit
into a carrier network for two or more connections. Unlike point-to-point,
which has exactly two sites connected to a single connection, multiple sites
privately connect into a carrier’s switched network to communicate with
each other. Circuit switching operates like a normal telephone call. ISDN is
an example of circuit-switched WAN technology.
• Packet switching: This type of transport is similar to circuit switching in
that multiple sites privately connect into a carrier-switched network.
However, packet switching involves the statistical multiplexing of packets
across shared circuits. Frame Relay, Multiprotocol Label Switching (MPLS),
broadband DSL and cable, and Metro Ethernet are all examples of packet
switching.
Some WAN technologies, such as Frame Relay and Asynchronous Transfer
Mode (ATM), use virtual circuits to ensure reliable communication between
two network devices. The two types of virtual circuits are switched virtual circuits
(SVC) and permanent virtual circuits (PVC). An SVC is dynamically
established on demand and is torn down when transmission is complete. A
connection uses SVCs when data transmission between devices is sporadic. A
PVC is a permanently established logical circuit and is useful for connections
between two devices in which data transfer is constant.
WAN dialup services are available as alternative backup technologies for traditional
WAN services. As the name implies, dialup services use plain old telephone
service (POTS) and are inexpensive (but relatively slow) alternatives
when the main WAN service goes down. Cisco routers offer two popular types
of dialup services: dial-on-demand routing (DDR) and dial backup. DDR can
be triggered automatically when the primary connection goes down or when a
traffic threshold is reached. Dial backup initiates a dial connection to another
router after it determines that the primary WAN service is unavailable. The
dial connection remains active until the WAN service returns

Integrated Services Digital Network

Integrated Services Digital Network (ISDN) is a set of technologies developed
to carry voice, video, and data across telephone networks. ISDN operates at
Layers 1 to 3 in the OSI reference model. ISDN was the first broadband service
deployed in the home. It operated at two to four times the speed of the
modem technologies of the day and provided “always-on” connectivity compared
to modem dialup. For many years ISDN received a lot of hype that it
never quite lived up to. Eventually it was dealt a death blow as far as home
use with the advent of DSL and high-speed cable services. However, it is still in
use in some businesses, so it’s worth a quick look here.

Frame Relay

Frame Relay is a packet-switched WAN service that operates at the physical
and logical layers of the OSI reference model. Frame Relay was originally
designed to operate over ISDN but today operates over a variety of network
interfaces. Typical communication speeds for Frame Relay are between 56
kbps and 2 Mbps (although lower and higher speeds are supported). Frame
Relay provides connection-oriented services using virtual circuits. A Frame
Relay virtual circuit is a logical connection between two data terminal equipment
(DTE) devices across a Frame Relay packet-switched network. A datalink
connection identifier (DLCI) uniquely identifies each virtual circuit. You
can multiplex multiple virtual circuits on a single physical circuit.
Frame Relay switched networks provide simple congestion-notification mechanisms.
Frame Relay switching equipment can mark a Frame Relay packet with
front-end congestion notification (FECN) or back-end congestion notification
(BECN). The equipment marks the packets with a FECN or BECN if congestion
occurs during the transmission of the packet. The DTE equipment at the
other end of a circuit notices whether a packet has experienced congestion and
notifies a higher layer that congestion has occurred. Additionally, the equipment
can mark a packet as discard eligible (DE) to indicate that it is less
important, which means that it can be dropped if congestion occurs.

ATM

ATM is a standard for cell-based relay that carries voice, video, and data in
small, fixed-size cells. ATM networks are connection-oriented networks that
combine the benefits of circuit switching (guaranteed capacity and constant
transmission delay) with those of packet switching (flexibility and efficiency for
intermittent traffic). ATM transmits at speeds from a few Mbps to many Gbps.
High-speed ATM circuits typically require optic-fiber cables to transmit such
high speeds. Speeds of these circuits are characterized as “Optical Carrier”
class and are represented as OC-number. The number represents the multiple of the base OC-1 standard circuit, which can carry 51.84 Mbps. Common
circuit speeds are OC-3 (155.52 Mbps), OC-12 (622.08 Mbps), and OC-192
(9953.28 Mbps, or roughly 10 Gbps).
Traditional circuit-based networks use time-division multiplexing (TDM), in
which users are assigned a predetermined time slot; no other device can transmit
during this time slot. If a station has a lot of data to send, it can transmit
only during its time slot, even if the other time slots are empty. Conversely, if
the station has nothing to transmit, the time slot is sent empty and is wasted.
This arrangement is called synchronous transmission.
ATM is asynchronous, meaning that time slots are available on demand. This
allows for a more efficient use of available bandwidth. ATM uses single-sized
cells (as opposed to the variable-sized frames in Frame Relay), which have 53
bytes. Computers usually define things in powers of 2 or 8. The 53-byte cell
size represents a compromise between the phone-standards folks and the datastandards
folks.
ATM networks have two devices: ATM switches and ATM endpoints. ATM
switches accept cells from an endpoint or another switch, evaluate the cell
header, and quickly forward the cell out another interface toward the destination.
An ATM endpoint contains an ATM network interface adapter and is
responsible for converting digital data into cells and back again. Examples of
ATM endpoints include workstations, LAN switches, routers, and video coderdecoders
(codecs).
ATM networks can mark traffic after it is converted from its original data format
to require different types of handling. Some traffic, such as voice and
video, must be transferred through the network at regular intervals with little
variation in delay. Otherwise, the destination receives low-quality voice or
video transmission. Data traffic is less sensitive to network delays and can be
handled differently.
To ensure the appropriate delivery for each of these traffic types, ATM devices
employ QoS mechanisms that involve reserving bandwidth, shaping traffic to
meet the reserved bandwidth, and policing traffic that exceeds the reservation.

MPLS

MPLS is a highly efficient WAN service that companies are quickly adopting
either as a replacement for legacy Frame Relay and ATM WANs or as a second
high-speed WAN service. MPLS is discussed in more depth in a later section.

Broadband

Increasingly, companies are leveraging cable, DSL, and other types of broadband
Internet services to deploy WAN services. They offer low-cost, high-bandwidth
connectivity that is often suitable for small branch office locations, such as
retail stores, small insurance office branches, and gas stations and convenience
stores. Broadband services are discussed in more depth in a later section.

Virtual Private Networks (VPN)

A VPN is a secured connection between two devices over a shared, unsecured
network. VPNs have been used for some time for mobile devices such as laptops
to connect to their corporate headquarters over the Internet. This is typically
called a remote-access VPN. Encryption provides security so that no one
else on the Internet can eavesdrop on the data being sent back and forth.
Increasingly, companies are taking advantage of VPNs to also connect branch
offices to headquarters locations over the Internet, called a site-to-site VPN.
Site-to-site VPNs can be a very cost-effective way to connect relatively small
locations to corporate headquarters over Internet services, such as broadband
cable and DSL. VPNs are also used to some degree to authenticate users to
local access points in a wireless environment. VPNs are addressed in a bit
more depth in a later section.

WAN Devices

Numerous types of devices are associated with WAN service delivery. The first
is a WAN switch. Usually located in a carrier’s network, a WAN switch is a
multiport internetworking device whose job is moving traffic from source to
destination. Routers at the customer sites attach to the edges of the carrier’s
switched network (for Frame Relay and ATM). WAN switches operate at
Layer 2, the data link layer, of the OSI model.
For many packet-switched services, often a WAN router is used at both the
access location, often called the Customer Premises Equipment (CPE), and
the nearest connectivity location of the WAN service provider, often called the
Point of Presence (PoP). Modern packet-switched services, such as MPLS,
broadband, and the Internet, rely on very large, very high-speed routers to
route traffic across the service provider network between PoPs. These routers
form the backbone of the modern Internet and global WAN connectivity services
and are sometimes called core routers. Routers sitting at the edges of the
network, providing WAN access to businesses, are often called edge routers.

Multiprotocol Label Switching (MPLS) Services

MPLS is a Layer 2 WAN backbone technology that delivers WAN and MAN
services, traffic engineering capabilities, and a converged network infrastructure
that can also be used to aggregate and transport Frame Relay, ATM, and
IP traffic. Originally developed by Cisco in the form of tag switching, MPLS
was adopted as an Internet standard by the Internet Engineering Task Force
(IETF). Service providers are the primary implementers of the technology.
Service providers offer MPLS services as an alternative to their traditional
Frame Relay, leased line, and ATM services. With MPLS networks, service
providers can offer services similar to traditional WAN technologies at lower
costs and provide additional IP-based services previously not available.
At the heart of MPLS is an encapsulation scheme that serves as an alternative
to traditional IP routing. When a packet comes into the service provider edge,
a router assigns a tag to the packet based on the destination IP network. The
tag is a type of shorthand for a traditional IP-based route. After the tag is
applied, the router forwards the packet into the MPLS core. The core routers
read the label, apply the appropriate services, and forward the packet based on
the label. As soon as the packet reaches the destination edge of the service
provider network, the MPLS label is removed, and the IP packet is forwarded
onto the IP network.
Traffic engineering is a core component for service providers that allows them
to deliver services predictably for each of their customers. MPLS traffic engineering
expands on the capabilities offered by ATM and Frame Relay. Tagged
IP packets are routed through the MPLS core based on the resources required
by the packet and available network resources. The MPLS network chooses
the shortest path for a traffic flow based on its resource requirements.
Resource requirements are determined by the size and priority of a traffic flow.
MPLS networks can honor IP QoS by delivering both best-effort delivery as
well as time and bandwidth-sensitive guarantees.
One of the MPLS services that service providers offer is virtual private networks.
Using MPLS labels, service providers can deliver IP-based services to
many customers without the complexity of traditional Frame Relay or ATM
circuit management. Customers can use private or public IP addressing without
concern about overlapping other customer addressing.
Another advantage of MPLS networks is any-to-any connectivity. Whereas in
Frame Relay and ATM networks, connections are point-to-point, MPLS services
allow customers to connect into the service and easily reach any other office
connected to the service. This removes some of the complexity of traffic engineering
that corporate customers would have to do themselves and allows the
service provider to offer an important value-added service as a WAN provider.
MPLS VPN services are as secure as Frame Relay in that one customer cannot
see the traffic from another customer even though they traverse the same
MPLS network. For additional security, customers can place firewalls between
their private network and the service providers, as well as encrypt the traffic as it
goes into the MPLS network. As long as the packets have standard IP headers,
the MPLS network can ship the packet to its destination.
Because MPLS networks look like a private intranet to the connected IP networks,
service providers can offer additional IP-based services such as QoS,
telephony support within the VPN, and centralized services such as web
hosting.

MPLS Labels

The forwarding mechanism in MPLS uses a label to decide where and how to send packets or cells through
the network. The label is applied at the ingress to the SP network and is removed at the network egress
point. The router responsible for adding the label is the only network router that needs to process the
entire packet header. The information contained in the header, along with the preconfigured instructions, is
used to generate the label. Labels can be based on IP destinations (this is what traditional routing uses) and
other parameters, such as IP sources, QoS, VPN membership, or specific routes for traffic engineering purposes.
MPLS is also designed to support forwarding mechanisms from other protocols. MPLS tags are 4
bytes or 32 bits long, which aids the speed at which the rest of the routers can process the forwarding
information (IP headers are much longer than that).

MPLS Security

An additional benefit of MPLS is a small measure
of security (as compared to Frame Relay or ATM).
As illustrated in the figure, as soon as the packet or
cell from a company enters the SP network, the
label assigned essentially keeps that packet segregated
from all other customers’ packets/cells.
Because there is no place where one customer can
view another customer’s packet/cells, there is no
danger of having someone outside the SP network
snoop for packets. Obviously this would not stop
someone bent on illegally accessing a company’s
information, but it does remove the possibility of
someone claiming that he “accidentally” received
the information. Unfortunately, the number of incidents
of people or groups intentionally stealing or
monitoring data has been on the rise over the past
several years. Because of this trend, many people
no longer consider MPLS to be “inherently
secure,” as it was once billed.
Many companies opt for encryption using technologies
such as IPsec (IP Security) to provide data
Although it is possible to encrypt MPLS, it is not
encrypted by default.
How Does the Router Know Where
to Send Stuff?
The routers in an MPLS network forward packets
based on labels, but the router must know the
relationship between a label and path through the
network. This relationship is established and communicated
throughout the network using
Forwarding Equivalence Classes (FEC). A FEC is a
specific path through the network of LSRs and is
equal to the destination network, stored in an IP
routing table. The LSRs simply look at the label
and forward the packet based on the contents of
the FEC. This is much simpler, faster, and more
flexible than traditional IP routing. Sometimes a
packet arrives at a router without a label (if it
security for their traffic traversing MPLS networks
(and, in general, any WAN type). This is especially
true where companies have offices with connections
in developing and emerging countries, where
the trust level of in-country providers may be
lower than in the U.S. and Europe.

MPLS Architecture

MPLS is divided into two layers or planes, each
having a specific function in the network. The layers
are the Control plane and the Data plane. The
Control plane is responsible for the exchange of
routing information (including labels) between
adjacent devices. The Data plane handles forwarding
operations.
It is important to note that MPLS is no more or
less secure than Frame or ATM. Also, there is a
common misconception that MPLS is encrypted.Although it is possible to encrypt MPLS, it is not
encrypted by default.

How Does the Router Know Where
to Send Stuff?

The routers in an MPLS network forward packets
based on labels, but the router must know the
relationship between a label and path through the
network. This relationship is established and communicated
throughout the network using
Forwarding Equivalence Classes (FEC). A FEC is a
specific path through the network of LSRs and is
equal to the destination network, stored in an IP
routing table. The LSRs simply look at the label
and forward the packet based on the contents of
the FEC. This is much simpler, faster, and more
flexible than traditional IP routing. Sometimes a
packet arrives at a router without a label (if itcame from a non-MPLS network). When this happens,
it is the router’s job to add a label so that the
packet can be properly forwarded through the
MPLS network.

Routing

Routers

Whereas switches and bridges operate at OSI Layer 2 (the data link layer),
routers primarily operate at OSI Layer 3 (the network layer). Like bridging,
the primary act of routing involves moving packets across a network from a
source to a destination. The difference involves the information that is used to
make the forwarding decisions. Routers make decisions based on network
layer protocols such as Internet Protocol (IP) and Novell NetWare
Internetwork Packet Exchange (IPX).
Routing gained popularity in the mid- to late 1980s as a result of internetworks
growing beyond the capability of bridges. Before this popularity, networks
were relatively small and isolated, and bridges could handle the jobs of
forwarding and segmentation. However, as networks grew, routers facilitated
larger scaling and more intelligent growth across wider physical distances.
Although routers are more expensive and complex than bridges, routing is the
core of the Internet today.

Routers Talk Among Themselves to Find Routes

Routing involves two processes: determining optimal routing paths through a
network, and forwarding packets along those paths. Routing algorithms make
the optimal path determination. As they determine routes, tables on the router
store the information. Routers communicate with each other and maintain
their routing tables through the exchange of messages over the network.
Routing updates are one particular type of message. A routing update contains
all or part of another router’s routing table and allows each router to build a
detailed picture of the overall network topology.
Routing algorithms fill routing tables with various types of information. The
primary piece of information relevant to routing is the next hop. Next-hop
associations tell a router that it can reach a particular destination by sending a
packet to a particular router representing the next hop on the way to its final
destination.
The process of exchanging information between routers is done using a routing
protocol. Put simply, a routing protocol is a series of messages that routers
use to exchange information about whether particular links are up or down,
and about other next-hop routers in the network. Three of the most common
routing protocols in use today are Open Shortest Path First (OSPF), Enhanced
Interior Gateway Routing Protocol (EIGRP), and Border Gateway Protocol
(BGP).

Routers Route Packets

When a router receives a packet, it attempts to associate the destination network
address in the packet to an appropriate next hop in its routing table. In
addition to next-hop associations, routers store other pertinent information in
routing tables. For multiple paths to a destination, a routing table might contain
information that allows the router to determine the desirability of one
path over another.
After a router determines an optimal path for a packet, it must forward the
packet toward the destination. The process of a router moving a packet from
its received port to the outgoing destination port is called switching. Although
the process of switching a packet on a router is similar to that of a Layer 2
switch, the decision criteria and the actual handling of the packet are different.
When a computer determines that it must send a packet to another host, it
specifies in the packet how to reach the destination network. If the destination
is unknown, the router typically drops the packet. If the destination is known,
the router changes the destination physical address in the packet to contain
that of the next hop. The router then transmits the packet out the destination
interface.
The next hop can be either the final destination or another router. Each router in
the process performs the same operation. As the packet moves through the network,
each router modifies the physical address stored in the packet but leaves
the network address untouched (because it determines the final destination).

Routers Bridge and Switches Route

In an ideal world, everything does what it is defined to do. This is not the case
with network devices. Routers can provide bridging functionality, and switches
are quickly becoming the high-density port, high-speed router of the campus.
Network devices, including switches and routers, make forwarding decisions
on OSI layers higher than the network layer. For example, routers can provide
firewall functionality in which the router inspects Layer 4 packet information,
and switches such as content switches can perform forwarding decisions based
on Layer 5 through 7 packet information (such as the URL in an HTTP packet).

Why Should I Care About Routing?

Routing is one of the fundamental aspects of networking.
Routers can learn possible routes (rather
than having to have the route manually configured
and constantly updated). This is one of the primary
reasons that ARPANET, which originally connected
seven sites, was able to scale exponentially into
the modern Internet in only a few years. Many
routers today have address tables with more than
100,000 entries, and they are updated constantly.

What Problems Need to Be Solved?

Routed networks are often large and complex, and
it would be prohibitively difficult to manage and
update network information on all routers all the
time. To account for this, several algorithms have
been developed. These algorithms allow the routers
to learn about the network and then make decisions
based on that information.
To learn paths (or routes) through a network and
then decide where to send packets, a router must
know the following information:
• Destination address: This is typically the IP
address of the data’s (packet) destination.
• Source address: The router also needs to know
where the information came from (typically an
IP address).
• Possible routes: These are likely routes that can
get information from the present location or
source to some other location (the destination or
closest known point).
• Best route: Routers also must know the best
(“best” can mean many things) path to the
intended destination.
• Status of routes: Routers also keep track of the
current state of routes to ensure timely delivery
of information.

What Exactly Does “Best” Mean?

Routers often make decisions about the best possible
path to get information from a source to a destination.
“Best” is loosely defined; it depends on
what the network values. These measurements of
value are called metrics. The network administrator
determines which metrics the network values.
Here are several metrics:
• Hop count: How many times a packet goes
through a router.
• Delay: The amount of time required to reach the
destination.
• Reliability: The bit error rate of each network
link.
• Maximum Transmission Unit (MTU): The maximum
message length (or packet size) allowed
on the path.
• Cost: An arbitrary value based on a network
administrator-determined value. Usually some
combination of other metrics.

Static Versus Dynamic

Routers must learn about the network around
them to determine where to send packets. This
information can be either manually entered (static
routes) or learned from other routers in the network
(dynamic routes):
• Static routes: When a network administrator
manually enters information about a route, it is
considered a static route. This information can
be changed only by a network administrator (in
other words, the router doesn’t learn about network
events). Static routes allow for tight control
of packets but become difficult to maintain
and are prone to human error. However, static
routes may be used to work around a temporary
problem or for performance enhancement.
• Dynamic routes: Routers on a network can
learn about possible routes and current route
status from other routers in the network. Routes
learned in this way are called dynamic routes.
Unlike static routes, any changes in the network
are learned without administrative intervention
and are automatically propagated throughout
the network.

Flat Versus Hierarchical

With flat networks, all routers must keep track of
all other routers on the network. As networks
grow, the amount of information contained in the
routing tables becomes larger and largerAlthough this method is simple, it can result in poor network performance as the
number of routing updates grows exponentially with each new router.
Hierarchical networks segment routers into logical groupings. This simplifies
routing tables and greatly reduces overheard traffic.
Intradomain Versus Interdomain
Intradomain and interdomain routing can be easily understood in the context of
large-scale hierarchical networks. In this regard, think of each segment as its
own autonomous network. Within each autonomous network, intradomain
routing protocols (also called Interior Gateway Protocols [IGP]) are used to
exchange routing information and forward packets. Interdomain routing protocols
(also called Exterior Gateway Protocols [EGP]) are used between
autonomous networks. This distinction is made because there are different performance
requirements for internal and external networks.

Distance Vector Versus Link-State

The two main classes of routing are distance vector routing and link-state routing.
With distance vector routing, also called “routing by rumor,” routers share their
routing table information with each other. Each router provides and receives
updates from its direct neighbor. In the figure, Router B shares information with
Routers A and C. Router C shares routing information with Routers B and D. A
distance vector describes the direction (port) and the distance (number of hops
or other metric) to some other router. When a router receives information from
another router, it increments whatever metric is being used. This process is called
distance accumulation. Routers using this method know the distance between
any two points in the network, but they do not know the exact topology of an
internetwork.

How Information Is Discovered with
Distance Vectors

Network discovery is the process of learning about
nondirectly connected routers. During network discovery,
routers accumulate metrics and learn the
best paths to various destinations in the network.
In the figure, each directly connected network has
a distance of 0. Router A learns about other networks
based on information it receives from
Router B. Routers increment the distance metric
for any route learned by an adjacent router. In
other words, Router A increments by 1 any distance
information it learns about other routers
from Router B.
As more time goes by, the router learns more
about the network and can process packets
more efficiently.

Link-State Routing

With link-state routing (also known as shortest
path first [SPF]), each router maintains a database
of topology information for the entire network.
Link-state routing provides better scaling than
distance-vector routing, because it sends updates
only when there is a change in the network, and
then it sends only information specific to the
change that occurred. Distance vector uses regular
updates and sends the whole routing table every
time. Link-state routing also uses a hierarchical
model, limiting the scope of route changes that
occur.

Lan Switching

Fast Computers Need Faster Networks

The PC emerged as the most common desktop computer in the 1980s. LANs emerged as a way to network PCs in a common location. Networking technologies such as Token Ring and Ethernet allowed users to share resources such as printers and exchange files with each other. As originally defined, Ethernet and Token Ring provided network access to multiple devices on the
same network segment or ring. These LAN technologies have inherent limitations as to how many devices can connect to a single segment, as well as the physical distance between computers. Desktop computers got faster, the number of computers grew, operating systems began multitasking (allowing multiple tasks to operate at the same time), and applications became more networkcentric. All these advancements resulted in congestion on LANs. To address these issues, two device types emerged: repeaters and bridges. Repeaters are simple Open Systems Interconnection (OSI) Layer 1 devices that allow networks to extend beyond their defined physical distances (which were limited to about 150 feet without the use of a repeater). Bridges are OSI Layer 2 devices that physically split a segment into two and reduce the amount of traffic on either side of the bridge. This setup allows more devices to connect to the LAN and reduces congestion. LAN switches emerged as a natural extension of bridging, revolutionizing the concept of local-area networking.

Switching Basics: It’s a Bridge

Network devices have one primary purpose: to pass network traffic from one segment to another. (There are exceptions, of course, such as network analyzers, which inspect traffic as it goes by.) With devices that independently make forwarding decisions, traffic can travel from its source to the destination. The higher up the OSI model a device operates, the deeper it looks into a packet to make a forwarding decision. Railroad-switching stations provide a similar example. The switches enable a train to enter the appropriate tracks (path) that take it to its final destination. If the switches are set wrong, a train can end up traveling to the wrong destination or traveling in a circle. Switching technology emerged as the replacement for bridging. Switches provide all the features of traditional bridging and more. Compared to bridges, switches provide superior throughput performance, higher port density, and lower per-port cost. The different types of bridging include the following:

• Transparent bridging primarily occurs in Ethernet networks.
• Source-route bridging occurs in Token Ring networks.
• Translational bridging occurs between different media. For example, a translational bridge might connect a Token Ring network to an Ethernet network.

Bridging and switching occur at the data link layer (Layer 2 in the OSI model), which means that bridges control data flow, provide transmission error handling, and enable access to physical media. Basic bridging is not complicated: A bridge or switch analyzes an incoming frame, determines where to forward the frame based on the packet’s header information (which contains information on the source and destination addresses), and forwards the frame toward
its destination. With transparent bridging, forwarding decisions happen one hop (or network segment) at a time. With source-route bridging, the frame contains a predetermined path to the destination. Bridges and switches divide networks into smaller, self-contained units. Because only a portion of the traffic is forwarded, bridging reduces the overall traffic that devices see on each connected network. The bridge acts as a kind of firewall in that it prevents frame-level errors from propagating from one segment to another. Bridges also accommodate communication among more devices than are supported on a single segment or ring. Bridges and switches essentially extend the effective length of a LAN, permitting more workstations to communicate with each other within a single broadcast domain. The primary difference between switches and bridges is that bridges segment a LAN into a few smaller segments. Switches, through their increased port density and speed, permit segmentation on a much larger scale. Modern-day switches used in corporate networks have hundreds of ports per chassis (unlike the four-port box connected to your cable modem).Additionally, modern-day switches interconnect LAN segments operating at different speeds. Switching describes technologies that are an extension of traditional bridges. Switches connect two or more LAN segments and make forwarding decisions about whether to transmit packets from one segment to another. When a
frame arrives, the switch inspects the destination and source Media Access Control (MAC) addresses in the packet. (This is an example of store-andforward switching.) The switch places an entry in a table indicating that the source MAC address is located off the switch interface on which the packet arrived. The bridge then consults the same table for an entry for the destination MAC address. If it has an entry for the destination MAC address, and the entry indicates that the MAC address is located on a different port from which the packet was received, the switch forwards the frame to the specified port. If the switch table indicates that the destination MAC address is located on the same interface on which the frame was just received, the bridge does not forward the frame. Why send it back onto the network segment from which item came? This decision is where a switch reduces network congestion. Finally, if the destination MAC address is not in the switch’s table, this indicates that the switch has not yet seen a frame destined for this MAC address. In this case, the switch then forwards the frames out all other ports (called flooding) except the one on which the packet was received. At their core, switches are multiport bridges. However, switches have radically matured into intelligent devices, replacing both bridges and hubs. Switches not only reduce traffic through the use of bridge tables, but also offer new functionality that supports high-speed connections, virtual LANs, and even traditional routing.

Ethernet

With the fundamentals of networking under our belt, we can now take a closer look at the infrastructure that makes up the networks we all use. This section focuses on the switches and routers that make up networks, along with the protocols that drive them. We start this section with a discussion of the Ethernet protocol, which defines the rules and processes by which computers in a local area communicate. Long before the Internet was in use, computers communicated locally using the Ethernet protocol, and it is still widely used. We then move on to local-area network (LAN) switching, an extension of the Ethernet protocol required when there are more computers in a local segment than can communicate efficiently. Switching is one of the core technologies in networking. One of the necessities in networking is link redundancy, something that makes it more likely that data reaches its intended target. Sometimes, however, link redundancy can create loops in the network, which causes an explosion of administrative traffic that can take down a network in a matter of minutes. Spanning Tree is one of the mechanisms that keeps these “broadcast storms” from wiping out your local network, so we look at how this important protocol works. We end this section with routing, which provides the basis for network communication over long distances. The advent of routing allowed the growth of the Internet and corporate networking as we know it....

History of Ethernet

Robert Metcalfe developed Ethernet at the famous Xerox Palo Alto Research Center (PARC) in 1972. The folks at Xerox PARC had developed a personal workstation with a graphical user interface. They needed a technology to network these workstations with their newly developed laser printers. (Remember, the first PC, the MITS altair, was not introduced to the public until 1975.) Metcalfe originally called this network the Alto Aloha Network. He changed the name to Ethernet in 1973 to make it clear that any type of device could connect to his network. He chose the name “ether” because the network carried bits to every workstation in the same manner that scientists once thought waves were propagated through space by the “luminiferous ether.”
Metcalfe’s first external publication concerning Ethernet was available to the public in 1976. Metcalfe left Xerox, and in 1979 he got Digital Equipment Corporation (DEC), Intel, and Xerox to agree on a common Ethernet standard called DIX. In 1982, the Institute of Electrical and Electronic Engineers (IEEE) adopted a standard based on Metcalfe’s Ethernet. Ethernet took off in academic networks and some corporate networks. It was cheap, and public domain protocols such as Internet Protocol (IP) ran natively on it. However, another company (IBM) wanted the world to adopt its protocol instead, called Token Ring. Before switching was introduced, Ethernet was more difficult to troubleshoot than Token Ring. Although Ethernet was less expensive to implement, larger corporations chose Token Ring because of their relationship with IBM and the ability to more easily troubleshoot problems. Early Ethernet used media such as coaxial cable, and a network could literally be a single long, continuous segment of coax cable tied into many computers. (This cable was known as Thinnet or Thicknet, depending on the thickness of the coax used.) When someone accidentally kicked the cable under his or her desk, this often produced a slight break in the network. A break meant that no one on the network could communicate, not just the poor schmuck who kicked the cable. Debugging usually entailed crawling under desks and following the cable until the break was found. In contrast, Token Ring had more sophisticated tools (than crawling on your knees) for finding the breaks. It was usually pretty obvious where the token stopped being passed and, voilà, you had your culprit. The battle for the LAN continued for more than ten years, until eventually Ethernet became the predominant technology. Arguably, it was the widespread adoption of Ethernet switching that drove the final nail in Token Ring’s coffin. Other LAN technologies, such as AppleTalk and Novell IPX, have been and continue to be introduced, but Ethernet prevails as the predominant technology for local high-speed connectivity. Thankfully, we have left behind early media such as coax for more sophisticated technologies.

What Is Ethernet?

Ethernet describes a system that links the computers in a building or within a local area. It consists of hardware (a network interface card), software, and cabling used to connect the computers. All computers on an Ethernet areattached to a shared data link, as opposed to traditional point-to-point networks, in which a single device connects to another single device. Because all computers share the same data link on an Ethernet network, the network needs a protocol to handle contention if multiple computers want to transmit data at the same time, because only one can talk at a time without causing interference. Metcalfe’s invention introduced the carrier sense multiple access collision detect (CSMA/CD) protocol. CSMA/CD defines how a computer should listen to the network before transmitting. If the network is quiet, the computer can transmit its data. However, a problem arises if more than one computer listens, hears silence, and transmits at the same time: The data collides. The collision-detect part of CSMA/CD defines a method in which transmitting computers back off when collisions occur and randomly attempt to restart transmission. Ethernet originally operated at 3 Mbps, but today it operates at speeds ranging from 10 Mbps (that’s 10 million bits per second) to 10 Gbps (that’s 10 billion bits per second).

Evolution of Ethernet

When Metcalfe originally developed Ethernet, computers were connected to a single copper cable. The physical limitations of a piece of copper cable carrying electrical signals restricted how far computers could be from each other on an Ethernet. Repeaters helped alleviate the distance limitations. Repeaters are small devices that regenerate an electrical signal at the original signal strength. This process allows an Ethernet to extend across an office floor that might exceed the Ethernet distance limitations. The addition or removal of a device on the Ethernet cable disrupts the network for all other connected devices. A device called an Ethernet hub solves this problem. First, each port on a hub is actually a repeater. Second, hubs let computers insert or remove themselves nondisruptively from the network. Finally, hubs simplify Ethernet troubleshooting and administration. As networks grow larger, companies need to fit more and more computers onto an Ethernet. As the number of computers increases, the number of collisions on the network increases. As collisions increase, useful network traffic decreases (administrative traffic actually increases because of all the error messages getting passed around). Networks come to a grinding halt when too many collisions occur. Ethernet bridges resolve this problem by physically breaking an Ethernet into two or more segments. This arrangement means that devices communicating on one side of the bridge do not collide with devices communicating on the other side of the bridge. Bridges also learn which devices are on each side and only transfer traffic to the network containing the destination device. A twoport bridge also doubles the bandwidth previously available, because each port is a separate Ethernet. Ethernet bridges evolved to solve the problem of connecting Ethernet networks to Token Ring networks. This process of translating a packet from one LAN technology to another is called translational bridging. As Ethernet networks continue to grow in a corporation, they become more complex, connecting hundreds and thousands of devices. Ethernet switches allow network administrators to dynamically break their networks into multiple Ethernet segments. Initially, switches operated as multiport Ethernet bridges. But eventually, as the cost per port decreased significantly, Ethernet switches replaced hubs, in which each connected device receives its own dedicated Ethernet bandwidth. With switches, collisions are no longer an issue, because connections between computer and switch can be point-to-point, and the Ethernet can both send and receive traffic at the same time. This ability to send and receive simultaneously is called full duplex, as opposed to traditional Ethernet, which operated at half duplex. Half duplex means that a device can receive or transmit traffic on the network, but not at the same time. If both happen at the same time, a collision occurs. This is different from subnetting in a couple of distinct ways. First, Ethernet is a Layer 2 protocol, and subnetting has to do with IP addressing (which is a Layer 3 function). Second, IP addressing is a logical segmentation scheme, and switching is a physical separation, because each end station has a dedicated physical port on the switch.

Address, Port, Pat and Nat

Addressing

Physical Addressing

In computing, a physical address, also real address, or binary address, is the memory address that is electronically (in the form of binary number) presented on the computer address bus circuitry in order to enable the data bus to access a particular storage cell of main memory.

In a computer with virtual memory, the term physical address is used mostly to differentiate from a virtual address. In particular, in computers utilizing memory management unit (MMU) to translate memory addresses, the virtual and physical address refer to address before and after MMU translation, respectively.

In computer networking, physical address is sometimes a synonym of MAC address. The address is actually used on network's data link layer, not on physical layer, as the name would suggest.

Note: There are two basic types of physical addresses when referencing Ethernet which are large and fixed physical addresses and proNET, which has small relatively easy to configure addresses.

Unaligned addressing

Depending upon its underlying computer architecture, the performance of a computer may be hindered by unaligned access to memory. As an example, a 16 bit computer with a 16 bit memory data bus such as an Intel 8086 generally works most efficiently if the access is aligned to an even address. In that case fetching one 16 bit value requires a single memory read operation, a single transfer over a data bus. Obviously, if the 16 bit data value starts at an odd address, the processor may actually need to perform two memory read cycles to load the value into it, i.e. one for the low address (throwing half of it away) and then a second to load the high address (again throwing half of the retrieved data away).

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Logical Address

In computer architectures, a logical address is the address at which a memory location appears to reside from the perspective of an executing application program. This may be different from the physical address due to the operation of a memory management unit (MMU) between the CPU and the memory bus. Physical memory may be mapped to different logical addresses for various purposes. For example, the same physical memory may appear at two logical addresses and if accessed by the program at one address, data will pass through the processor cache whereas if it is accessed at the other address, it will bypass the cache.

In a system supporting virtual memory, there may actually not be any physical memory mapped to a logical address until an access is attempted. The access triggers special functions of the operating system which reprogram the MMU to map the address to some physical memory, perhaps writing the old contents of that memory to disk and reading back from disk what the memory should contain at the new logical address. In this case, the logical address may be referred to as a virtual address.

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Port Address Translation

Port Address Translation (PAT) is a feature of a network device that translates TCP or UDP communications made between hosts on a private network and hosts on a public network. It allows a single public IP address to be used by many hosts on a private network, which is usually a Local Area Network or LAN.

A PAT device transparently modifies IP packets as they pass through it. The modifications make all the packets which it sends to the public network from the multiple hosts on the private network appear to originate from a single host, (the PAT device) on the public network.

Relationship between NAT and PAT

PAT is a subset of NAT, and is closely related to the concept of Network Address Translation. PAT is also known as NAT Overload. In PAT there is generally only one publicly exposed IP address and multiple private hosts connecting through the exposed address. Incoming packets from the public network are routed to their destinations on the private network by reference to a table held within the PAT device which keeps track of public and private port pairs.

In PAT, both the sender's private IP and port number are modified; the PAT device chooses the port numbers which will be seen by hosts on the public network. In this way, PAT operates at layer 3 (network) and 4 (transport) of the OSI model, whereas basic NAT only operates at layer 3.

Establishing Two-Way Communication

Every TCP and UDP packet contains both a source IP address and source port number as well as a destination IP address and destination port number. These four pieces of information, taken together, form a socket.

For publicly accessible services such as web servers and mail servers the port number is important. For example, port 80 connects to the web server software and port 25 to a mail server's SMTP daemon. The IP address of a public server is also important, similar in global uniqueness to a postal address or telephone number. Both IP address and port must be correctly known by all hosts wishing to successfully communicate.

Private IP addresses as described in RFC 1918 are significant only on private networks where they are used, which is also true for host ports. Ports are unique endpoints of communication on a host, so a connection through the PAT device is maintained by the combined mapping of port and IP address.

PAT resolves conflicts that would arise through two different hosts using the same source port number to establish unique connections at the same time.

An Analogy of PAT

A PAT device is similar to the receptionist at an office that has one public telephone number. Outbound phone calls made from the office all appear to come from the same telephone number. However, incoming calls have to be transferred to the correct private extension by an operator asking the caller who they'd like to speak with; private extensions cannot be dialed directly from outside.

Translation of the Endpoint

With PAT, all communication sent to external hosts actually contain the external IP address and port information of the PAT device instead of internal host IPs or port numbers.

• When a computer on the private (internal) network sends a packet to the external network, the PAT device replaces the internal IP address in the source field of the packet header (sender's address) with the external IP address of the PAT device. It then assigns the connection a port number from a pool of available ports, inserting this port number in the source port field (much like the post office box number), and forwards the packet to the external network. The PAT device then makes an entry in a translation table containing the internal IP address, original source port, and the translated source port. Subsequent packets from the same connection are translated to the same port number.

• The computer receiving a packet that has undergone PAT establishes a connection to the port and IP address specified in the altered packet, oblivious to the fact that the supplied address is being translated (analogous to using a post office box number).

• A packet coming from the external network is mapped to a corresponding internal IP address and port number from the translation table, replacing the external IP address and port number in the incoming packet header (similar to the translation from post office box number to street address). The packet is then forwarded over the inside network. Otherwise, if the destination port number of the incoming packet is not found in the translation table, the packet is dropped or rejected because the PAT device doesn't know where to send it.

PAT will only translate IP addresses and ports of its internal hosts, hiding the true endpoint of an internal host on a private network.

Visibility of Operation

The PAT operation is typically transparent to both the internal and external hosts.

Typically the internal host is aware of the true IP address and TCP or UDP port of the external host. Typically the PAT device may function as the default gateway for the internal host. However the external host is only aware of the public IP address for the PAT device and the particular port being used to communicate on behalf of a specific internal host.

Uses of PAT

Software firewalls and broadband network access devices (e.g. ADSL routers) are examples of network technologies that may contain PAT implementations. When configuring these devices, the external network is the Internet and the internal network is a LAN.

Examples of PAT

A host at IP address 192.168.0.2 on the private network may ask for a connection to a remote host on the public network. The initial packet is given the address 192.168.0.2:15345. The PAT device (which we assume has a public IP of 1.2.3.4) may arbitrarily translate this source address:port pair to 1.2.3.4:16529, making an entry in its internal table that port 16529 being used for a connection by 192.168.0.2 on the private network. When a packet is received from the public network by the PAT device for address 1.2.3.4:16529 the packet is forwarded to 192.168.0.2:15345.

Advantages of PAT

In addition to the advantages provided by NAT:

• PAT allows multiple internal hosts to share a single external IP address.

Disadvantages of PAT

• Scalability - Many hosts on the private network make many connections to the public network. Since there are only a limited number of ports available, the PAT device may eventually have insufficient space in the translation table. While there are thousands of ports available, and they are recycled quickly, some network communications consume multiple ports nearly simultaneously in a single logical transaction (an HTTP request for a web page with many embedded objects; some VoIP applications). Sufficiently-large LANs that frequently sustain this type of traffic could periodically consume all available ports.
• Firewall complexity - Because the inside addresses are all disguised behind one publicly-accessible address, it is impossible for outside machines to initiate a connection to a particular inside machine without special configuration on the firewall to forward connections to a particular port. This has a considerable impact upon applications such as VOIP, videoconferencing, and other peer-to-peer applications.

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TCP/IP

TCP/IP

Introduction to TCP/IP

Summary: TCP and IP were developed by a Department of Defense (DOD) research project to connect a number different networks designed by different vendors into a network of networks (the "Internet"). It was initially successful because it delivered a few basic services that everyone needs (file transfer, electronic mail, remote logon) across a very large number of client and server systems. Several computers in a small department can use TCP/IP (along with other protocols) on a single LAN. The IP component provides routing from the department to the enterprise network, then to regional networks, and finally to the global Internet. On the battlefield a communications network will sustain damage, so the DOD designed TCP/IP to be robust and automatically recover from any node or phone line failure. This design allows the construction of very large networks with less central management. However, because of the automatic recovery, network problems can go undiagnosed and uncorrected for long periods of time.

As with all other communications protocol, TCP/IP is composed of layers:

• IP - is responsible for moving packet of data from node to node. IP forwards each packet based on a four byte destination address (the IP number). The Internet authorities assign ranges of numbers to different organizations. The organizations assign groups of their numbers to departments. IP operates on gateway machines that move data from department to organization to region and then around the world.
• TCP - is responsible for verifying the correct delivery of data from client to server. Data can be lost in the intermediate network. TCP adds support to detect errors or lost data and to trigger retransmission until the data is correctly and completely received.
• Sockets - is a name given to the package of subroutines that provide access to TCP/IP on most systems.

Network of Lowest Bidders

The Army puts out a bid on a computer and DEC wins the bid. The Air Force puts out a bid and IBM wins. The Navy bid is won by Unisys. Then the President decides to invade Grenada and the armed forces discover that their computers cannot talk to each other. The DOD must build a "network" out of systems each of which, by law, was delivered by the lowest bidder on a single contract.

The Internet Protocol was developed to create a Network of Networks (the "Internet"). Individual machines are first connected to a LAN (Ethernet or Token Ring). TCP/IP shares the LAN with other uses (a Novell file server, Windows for Workgroups peer systems). One device provides the TCP/IP connection between the LAN and the rest of the world.

To insure that all types of systems from all vendors can communicate, TCP/IP is absolutely standardized on the LAN. However, larger networks based on long distances and phone lines are more volatile. In the US, many large corporations would wish to reuse large internal networks based on IBM's SNA. In Europe, the national phone companies traditionally standardize on X.25. However, the sudden explosion of high speed microprocessors, fiber optics, and digital phone systems has created a burst of new options: ISDN, frame relay, FDDI, Asynchronous Transfer Mode (ATM). New technologies arise and become obsolete within a few years. With cable TV and phone companies competing to build the National Information Superhighway, no single standard can govern citywide, nationwide, or worldwide communications.

The original design of TCP/IP as a Network of Networks fits nicely within the current technological uncertainty. TCP/IP data can be sent across a LAN, or it can be carried within an internal corporate SNA network, or it can piggyback on the cable TV service. Furthermore, machines connected to any of these networks can communicate to any other network through gateways supplied by the network vendor.

Addresses

Each technology has its own convention for transmitting messages between two machines within the same network. On a LAN, messages are sent between machines by supplying the six byte unique identifier (the "MAC" address). In an SNA network, every machine has Logical Units with their own network address. DECNET, Appletalk, and Novell IPX all have a scheme for assigning numbers to each local network and to each workstation attached to the network.

On top of these local or vendor specific network addresses, TCP/IP assigns a unique number to every workstation in the world. This "IP number" is a four byte value that, by convention, is expressed by converting each byte into a decimal number (0 to 255) and separating the bytes with a period. For example, the PC Lube and Tune server is 130.132.59.234.

An organization begins by sending electronic mail to Hostmaster@INTERNIC.NET requesting assignment of a network number. It is still possible for almost anyone to get assignment of a number for a small "Class C" network in which the first three bytes identify the network and the last byte identifies the individual computer. The author followed this procedure and was assigned the numbers 192.35.91.* for a network of computers at his house. Larger organizations can get a "Class B" network where the first two bytes identify the network and the last two bytes identify each of up to 64 thousand individual workstations. Yale's Class B network is 130.132, so all computers with IP address 130.132.*.* are connected through Yale.

The organization then connects to the Internet through one of a dozen regional or specialized network suppliers. The network vendor is given the subscriber network number and adds it to the routing configuration in its own machines and those of the other major network suppliers.

There is no mathematical formula that translates the numbers 192.35.91 or 130.132 into "Yale University" or "New Haven, CT." The machines that manage large regional networks or the central Internet routers managed by the National Science Foundation can only locate these networks by looking each network number up in a table. There are potentially thousands of Class B networks, and millions of Class C networks, but computer memory costs are low, so the tables are reasonable. Customers that connect to the Internet, even customers as large as IBM, do not need to maintain any information on other networks. They send all external data to the regional carrier to which they subscribe, and the regional carrier maintains the tables and does the appropriate routing.

New Haven is in a border state, split 50-50 between the Yankees and the Red Sox. In this spirit, Yale recently switched its connection from the Middle Atlantic regional network to the New England carrier. When the switch occurred, tables in the other regional areas and in the national spine had to be updated, so that traffic for 130.132 was routed through Boston instead of New Jersey. The large network carriers handle the paperwork and can perform such a switch given sufficient notice. During a conversion period, the university was connected to both networks so that messages could arrive through either path.

Subnets

Although the individual subscribers do not need to tabulate network numbers or provide explicit routing, it is convenien

t for most Class B networks to be internally managed as a much smaller and simpler version of the larger network organizations. It is common to subdivide the two bytes available for internal assignment into a one byte department number and a one byte workstation ID.

The enterprise network is built using commercially available TCP/IP router boxes. Each router has small tables with 255 entries to translate the one byte department number into selection of a destination Ethernet connected to one of the routers. Messages to the PC Lube and Tune server (130.132.59.234) are sent through the national and New England regional networks based on the 130.132 part of the number. Arriving at Yale, the 59 department ID selects an Ethernet connector in the C& IS building. The 234 selects a particular workstation on that LAN. The Yale network must be updated as new Ethernets and departments are added, b

ut it is not effected by changes outside the university or the movement of machines within the department.

A Uncertain Path

Every time a message arrives at an IP router, it makes an individual decisio

n about where to send it next. There is concept of a session with a preselected path for all traffic. Consider a company with facilities in New York, Los Angeles, Chicago and Atlanta. It could build a network from four phone lines forming a loop (NY to Chicago to LA to Atlanta to NY). A message arriving at the NY router could go to LA via either Chicago or Atlanta. The reply could come back the other way.

How does the router make a decision between routes? There is no correct answer

. Traffic could be routed by the "clockwise" algorithm (go NY to Atlanta, LA to Chicago). The routers could alternate, sending one message to Atlanta and the next to Chicago. More sophisticated routing measures traffic patterns and sends data through the least busy link.

If one phone line in this network breaks down, traffic can still reach its destination through a roundabout path. After losing the NY to Chicago line, data can be sent NY to Atlanta to LA to Chicago. This provides continued service though with degraded performance. This kind of recovery is the primary design feature of IP. The loss of the line is immediately detected by the routers in NY and Chicago, but somehow this information must be sent to the other nodes. Otherwise, LA could continue to send NY messages through Chicago, where t

hey arrive at a "dead end." Each network adopts some Router Protocol which periodically updates the routing tables throughout the network with information about changes in route status.

If the size of the network grows, then the complexity of the routing updates will increase as will the cost of transmitting them. Building a single network that covers the entir

e US would be unreasonably complicated. Fortunately, the Internet is designed as a Network of Networks. This means that loops and redundancy are built into each regional carrier. The regional network handles its own problems and reroutes messages internally. Its Router Protocol updates the tables in its own routers, but no routing updates need to propagate from a regional carrier to the NSF spine or to the other regions (unless, of course, a subscriber switches permanently from one region to another).

Undiagnosed Problems

IBM designs its SNA networks to be centrally managed. If any error occurs, it is rep

orted to the network authorities. By design, any error is a problem that should be corrected or repaired. IP networks, however, were designed to be robust. In battlefield conditions, the loss of a node or line is a normal circumstance. Casualties can be sorted out later on, but the network must stay up. So IP networks are robust. They automatically (and silently) reconfigure themselves when something goes wrong. If there is enough redundancy built into the system, then communication is maintained.

In 1975 when SNA was designed, such redundancy would be prohibitively expensive, or it might have been argued that only the Defense Department could afford it. Today, however, simple routers cost no more than a PC. However, the TCP/IP design that, "Errors are normal and can be largely ignored," produces problems of its own.

Data traffic is frequently organized around "hubs," much like airline traffic. One could imagine an IP router in Atlanta routing messages for smaller cities throughout the Southeast. The problem is that data arrives without a reservation. Airline companies experience the problem around major events, like the Super Bowl. Just before the game, everyone wants to fly into the city. After the game, everyone wants to fly out. Imbalance occurs on the network wh

en something new gets advertised. Adam Curry announced the server at "mtv.com" and his regional carrier was swamped with traffic the next day. The problem is that messages come in from the entire world over high speed lines, but they go out to mtv.com over what was then a slow speed phone line.

Occasionally a snow storm cancels flights and airports fill up with stranded passengers. Many go off to hotels in town. When data arrives at a congested router, there is no place to send the overflow. Excess packets are simply discarded. It becomes the responsibility of the sender to retry the data a few seconds later and to persist until it finally gets through. This

recovery is provided by the TCP component of the Internet protocol.

TCP was designed to recover from node or line failures where the network propagates routing table changes to all router nodes. Since the update takes some time, TCP is slow to initiate recovery. The TCP algorithms are not tuned to optimally handle packet loss due to traffic congestion. Instead, the traditional Internet response to traffic problems has been to increase the speed of lines and equipment in order to say ahead of growth in demand.

TCP treats the data as a stream of bytes. It logically assigns a sequence number to each byte. The TCP packet has a header that says, in effect, "This packet starts with byte 379

642 and contains 200 bytes of data." The receiver can detect missing or incorrectly sequenced packets. TCP acknowledges data that has been received and retransmits data that has been lost. The TCP design means that error recovery is done end-to-end between the Client and Server machine. There is no formal standard for tracking problems in the middle of the network, though each network has adopted some ad hoc tools.

Need to Know

There are three levels of TCP/IP knowledge. Those who administer a regional o

r national network must design a system of long distance phone lines, dedicated routing devices, and very large configuration files. They must know the IP numbers and physical locations of thousands of subscriber networks. They must also have a formal network monitor strategy to detect problems and respond quickly.

Each large company or university that subscribes to the Internet must have an intermediate level of network organization and expertise. A half dozen routers might be configured to connect several dozen departmental LANs in several buildings. All traffic outside the organization would typically be routed to a single connection to a regional network provider.

However, the end user can install TCP/IP on a personal computer without any knowledge of either the corporate or regional network. Three pieces of information are required:

1. The IP address assigned to this personal computer
2. The part of the IP address (the subnet mask) that distinguishes other machines on the same LAN (messages can be sent to them directly) from machines in other depar tments or elsewhere in the world (which are sent to a router machine)
3. The IP address of the router machine that connects this LAN to the rest of the world.

In the case of the PCLT server, the IP address is 130.132.59.234. Since the first three bytes designate this department, a "subnet mask" is defined as 255.255.255.0 (255 is the largest byte value and represents the number with all bits turned on). It is a Yale convention (which we recommend to everyone) that the router for each department have station number 1 within the department network. Thus the PCLT router is 130.132.59.1. Thus the PCLT server is configured with the values:

• My IP address: 130.132.59.234
• Subnet mask: 255.255.255.0
• Default router: 130.132.59.1

The subnet mask tells the server that any other machine with an IP address beginning 130.132.59.* is on the same department LAN, so messages are sent to it directly. Any IP address beginning with a different value is accessed indirectly by sending the message through the router at 130.132.59.1 (which is on the departmental LAN).

P2P

Peer to Peer (P2P)

P2P networking has generated tremendous interest worldwide among both Internet surfers and computer networking professionals. P2P software systems like Kazaa and Napster rank amongst the most popular software applications ever. Numerous businesses and Web sites have promoted "peer to peer" technology as the future of Internet networking.

Although they have actually existed for many years, P2P technologies promise to radically change the future of networking. P2P file sharing software has also created much controversy over legality and "fair use." In general, experts disagree on various details of P2P and precisely how it will evolve in the future.

Traditional Peer to Peer Networks

The P2P acronym technically stands for peer to peer. Webopedia defines p2p as

"A type of network in which each workstation has equivalent capabilities and responsibilities. This differs from client/server architectures, in which some computers are dedicated to serving the others."

This definition captures the traditional meaning of peer to peer networking. Computers in a peer to peer network are typically situated physically near to each other and run similar networking protocols and software. Before home networking became popular, only small businesses and schools built peer to peer networks.

Home Peer to Peer Networks

Most home computer networks today are peer to peer networks. Residential users configure their computers in peer workgroups to allow sharing of files, printers and other resources equally among all of the devices. Although one computer may act as a file server or Fax server at any given time, other home computers often have equivalent capability to handle those responsibilities.

Both wired and wireless home networks qualify as peer to peer environments. Some may argue that the installation of a network router or similar centerpiece device means that network is no longer peer to peer. From the networking point of view, this is inaccurate. A router simply joins the home network to the Internet; it does not by itself change how resources within the network are shared.

P2P File Sharing Networks

When most people hear the term "P2P", they think not of traditional peer networks, but rather peer to peer file sharing over the Internet. P2P file sharing systems have become the single most popular class of Internet applications in this decade.

A P2P network implements search and data transfer protocols above the Internet protocol (IP) To access a P2P network, users simply download and install a suitable P2P client application.

Numerous P2P networks and P2P software applications exist. Some P2P applications work only with one P2P network, while others operate cross-network. Likewise, some P2P networks support only one application, while others support multiple applications.

OSI Layers

The Open Systems Interconnection Reference Model (OSI Reference Model or OSI Model) is an abstract description for layered communications and computer network protocol design. It was developed as part of the Open Systems Interconnection (OSI) initiative. In its most basic form, it divides network architecture into seven layers which, from top to bottom, are the Application, Presentation, Session, Transport, Network, Data-Link, and Physical Layers. It is therefore often referred to as the OSI Seven Layer Model.

A layer is a collection of conceptually similar functions that provide services to the layer above it and receives service from the layer below it. For example, a layer that provides error-free communications across a network provides the path needed by applications above it, while it calls the next lower layer to send and receive packets that make up the contents of the path.

Source - Wikipedia

History

In 1977, work on a layered model of network architecture was started, and the International Organization for Standardization (ISO) began to develop its OSI framework architecture. OSI has two major components: an abstract model of networking, called the Basic Reference Model or seven-layer model, and a set of specific protocols.

Note: The standard documents that describe the OSI model can be freely downloaded from the ITU-T as the X.200-series of recommendations. A number of the protocol specifications are also available as part of the ITU-T X series. The equivalent ISO and ISO/IEC standards for the OSI model are available from the ISO, but only some of the ISO/IEC standards are available as cost-free downloads.

All aspects of OSI design evolved from experiences with the CYCLADES network, which also influenced Internet design. The new design was documented in ISO 7498 and its various addenda. In this model, a networking system is divided into layers. Within each layer, one or more entities implement its functionality. Each entity interacts directly only with the layer immediately beneath it, and provides facilities for use by the layer above it.

Protocols enable an entity in one host to interact with a corresponding entity at the same layer in another host. Service definitions abstractly describe the functionality provided to an (N)-layer by an (N-1) layer, where N is one of the seven layers of protocols operating in the local host.

A common joke is that there exists an eighth layer titled "User", but there is no official standard dictating this.

Layer 7: Application Layer

Main article: Application Layer

The application layer is the OSI layer closest to the end user, which means that both the OSI application layer and the user interact directly with the software application. This layer interacts with software applications that implement a communicating component. Such application programs fall outside the scope of the OSI model. Application layer functions typically include identifying communication partners, determining resource availability, and synchronizing communication. When identifying communication partners, the application layer determines the identity and availability of communication partners for an application with data to transmit. When determining resource availability, the application layer must decide whether sufficient network resources for the requested communication exist. In synchronizing communication, all communication between applications requires cooperation that is managed by the application layer. Some examples of application layer implementations include Telnet, Hypertext Transfer Protocol (HTTP), File Transfer Protocol (FTP) , and Simple Mail Transfer Protocol (SMTP).

Layer 6: Presentation Layer

Main article: Presentation Layer

The Presentation Layer establishes a context between Application Layer entities, in which the higher-layer entities can use different syntax and semantics, as long as the Presentation Service understands both and the mapping between them. The presentation service data units are then encapsulated into Session Protocol Data Units, and moved down the stack.

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 original presentation structure used the Basic Encoding Rules of Abstract Syntax Notation One (ASN.1), with capabilities such as converting an EBCDIC-coded text file to an ASCII-coded file, or serializing objects and other data structures into and out of XML.

Layer 5: Session Layer

Main article: Session Layer

The Session Layer controls the dialogues (connections) between computers. It establishes, manages and terminates the connections between the local and remote application. It provides for full-duplex, half-duplex, or simplex operation, and establishes checkpointing, adjournment, termination, and restart procedures. The OSI model made this layer responsible for "graceful close" of sessions, which is a property of TCP, and also for session checkpointing and recovery, which is not usually used in the Internet Protocol Suite. The Session Layer is commonly implemented explicitly in application environments that use remote procedure calls (RPCs).

Layer 4: Transport Layer

The Transport Layer provides transparent transfer of data between end users, providing reliable data transfer services to the upper layers. The Transport Layer controls the reliability of a given link through flow control, segmentation/desegmentation, and error control. Some protocols are state and connection oriented. This means that the Transport Layer can keep track of the segments and retransmit those that fail.

Although not developed under the OSI Reference Model and not strictly conforming to the OSI definition of the Transport Layer, the best known examples of a Layer 4 protocol are the Transmission Control Protocol (TCP) and User Datagram Protocol (UDP).^{[citation needed]}

Of the actual OSI protocols, there are five classes of connection-mode transport protocols ranging from class 0 (which is also known as TP0 and provides the least error recovery) to class 4 (which is also known as TP4 and is designed for less reliable networks, similar to the Internet). Class 0 contains no error recovery, and was designed for use on network layers that provide error-free connections. Class 4 is closest to TCP, although TCP contains functions, such as the graceful close, which OSI assigns to the Session Layer. Also, all OSI TP connection-mode protocol classes provide expedited data and preservation of record boundaries, both of which TCP is incapable. Detailed characteristics of TP0-4 classes are shown in the following table:^[5]

Feature Name	TP0	TP1	TP2	TP3	TP4
Connection oriented network	Yes	Yes	Yes	Yes	Yes
Connectionless network	No	No	No	No	Yes
Concatenation and separation	No	Yes	Yes	Yes	Yes
Segmentation and reassembly	Yes	Yes	Yes	Yes	Yes
Error Recovery	No	Yes	No	Yes	Yes
Reinitiate connection (if an excessive number of PDUs are unacknowledged)	No	Yes	No	Yes	No
multiplexing and demultiplexing over a single virtual circuit	No	No	Yes	Yes	Yes
Explicit flow control	No	No	Yes	Yes	Yes
Retransmission on timeout	No	No	No	No	Yes
Reliable Transport Service	No	Yes	No	Yes	Yes

Perhaps an easy way to visualize the Transport Layer is to compare it with a Post Office, which deals with the dispatch and classification of mail and parcels sent. Do remember, however, that a post office manages the outer envelope of mail. Higher layers may have the equivalent of double envelopes, such as cryptographic presentation services that can be read by the addressee only. Roughly speaking, tunneling protocols operate at the Transport Layer, such as carrying non-IP protocols such as IBM's SNA or Novell's IPX over an IP network, or end-to-end encryption with IPsec. While Generic Routing Encapsulation (GRE) might seem to be a Network Layer protocol, if the encapsulation of the payload takes place only at endpoint, GRE becomes closer to a transport protocol that uses IP headers but contains complete frames or packets to deliver to an endpoint. L2TP carries PPP frames inside transport packet.

Layer 3: Network Layer

Main article: Network Layer

The Network Layer provides the functional and procedural means of transferring variable length data sequences from a source to a destination via one or more networks, while maintaining the quality of service requested by the Transport Layer. The Network Layer performs network routing functions, and might also perform fragmentation and reassembly, and report delivery errors. Routers operate at this layer—sending data throughout the extended network and making the Internet possible. This is a logical addressing scheme – values are chosen by the network engineer. The addressing scheme is hierarchical.

The best-known example of a Layer 3 protocol is the Internet Protocol (IP). It manages the connectionless transfer of data one hop at a time, from end system to ingress router, router to router, and from egress router to destination end system. It is not responsible for reliable delivery to a next hop, but only for the detection of errored packets so they may be discarded. When the medium of the next hop cannot accept a packet in its current length, IP is responsible for fragmenting the packet into sufficiently small packets that the medium can accept.

A number of layer management protocols, a function defined in the Management Annex, ISO 7498/4, belong to the Network Layer. These include routing protocols, multicast group management, Networ