DATA AND COMPUTER COMMUNICATIONS PDF
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NOTICE This manual contains solutions to all of the review questions and homework problems in Data and Computer Communications, Eighth Edition. Fundamentals of Data Communications: Part I, Chapters 8 (circuit switch the book in PDF (Adobe Acrobat) format, and sign-up information for. Appendix B Projects for Teaching Data and Computer Communications B Practical . PDF files: Reproductions of all figures and tables from the book.
CH 11 -DCC10e.pdf - Data and Computer Communications Tenth
Each Hz signal can be sampled at a rate of 1 kHz. If 4-bit samples are used, then each signal requires 4 kbps, for a total data rate of 16 kbps. This scheme will work only if the line can support a data rate of 16 kbps in a bandwidth of Hz.
In time-division multiplexing, the entire channel is assigned to the source for a fraction of the time. If there is spare bandwidth, then the incremental cost of the transmission can be negligible.
Data and Computer Communications, 8th Edition
The new station pair is simply added to an unused subchannel. If there is no unused subchannel it may be possible to redivide the existing subchannels creating more subchannels with less bandwidth. If, on the other hand, a new pair causes a complete new line to be added, then the incremental cost is large indeed.
What the multiplexer receives from attached stations are several bit streams from different sources.
What the multiplexer sends over the multiplexed transmission line is a bit stream from the multiplexer. As long as the multiplexer sends what can be interpreted as a bit stream to the demultiplexer at the other end, the system will work.
The multiplexer, for example, may use a self-clocking signal. The incoming stream may be, on the other hand, encoded in some other format.
The multiplexer receives and understands the incoming bits and sends out its equivalent set of multiplexed bits. In synchronous TDM, using character interleaving, the character is placed in a time slot that is one character wide.
The character is delimited by the bounds of the time slot, which are defined by the synchronous transmission scheme. Thus, no further delimiters are needed. When the character arrives at its destination, the start and stop bits can be added back if the receiver requires these. TDM's focus is on the medium rather than the information that travels on the medium.
Its services should be transparent to the user. It offers no flow or error control. These must be provided on an individual-channel basis by a link control protocol. The actual bit pattern is If a receiver gets out of synchronization it can scan for this pattern and resynchronize.
This pattern would be unlikely to occur in digital data. Analog sources cannot generate this pattern. It corresponds to a sine wave at 4, Hz and would be filtered out from a voice channel that is band limited. One SYN character, followed by 20 bit terminal characters, followed by stuff bits. The available capacity is 1.
This is a practical limit based on the performance characteristics of a statistical multiplexer. If the receiver is on the framing pattern no searching , the minimum reframe time is 12 frame times the algorithm takes 12 frames to decide it is "in frame".
Hence it must search the maximum number of bits 55 to find it. Each search takes 12T f. Assuming the system is random, the reframing is equally to start on any bit position. Hence on the average it starts in the middle or halfway between the best and worst cases. Therefore, the channel cost will be only one-fourth, since one channel rather than four is now needed.
The same reasoning applies to termination charges. The present solution requires eight low speed modems four pairs of modems. The new solution requires two higher-speed modems and two multiplexers. The reliability of the multiplexed solution may be somewhat less. The new system does not have the redundancy of the old system. A failure anywhere except at the terminals will cause a complete loss of the system. Each multiplexer also acts as a buffer.
It can accept bits in asynchronous form, buffer them and transmit them in synchronous form, and vice versa. Assume a continuous stream of STDM frames.
If a delimiter is used, bit or character-stuffing may be needed. Only a recipient who knows the spreading code can recover the encoded information.
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A receiver, hopping between frequencies in synchronization with the transmitter, picks up the message. Each user uses a different spreading code. The receiver picks out one signal by matching the spreading code. Thus, to achieve the desired SNR, the signal must be spread so that 56 KHz is carried in very large bandwidths.
Thus a far higher SNR is required without spread spectrum. Period of the PN sequence is 15 b. MFSK c. Same as for Problem 9. This is from the example in Section 6. We need three more sets of 8 frequencies. The second set can start at kHz, with 8 frequencies separated by 50 kHz each. The third set can start at kHz, and the fourth set at kHz.
The first generator yields the sequence: The second generator yields the sequence: Because of the patterns evident in the second half of the latter sequence, most people would consider it to be less random than the first sequence. See [KNUT98], page 13 for a discussion. As discussed in the answer to Problem 9. Now, if we use a linear congruential generator of the following form: Often, a and c are chosen to create a sequence of alternating even and odd integers.
The simulation depends on counting the number of pairs of integers whose greatest common divisor is 1. With truly random integers, one-fourth of the pairs should consist of two even integers, which of course have a gcd greater than 1. This never occurs with sequences that alternate between even and odd integers. For a further discussion, see Danilowicz, R. Subscriber line: Two stations of different data rates can exchange packets because each connects to its node at its proper data rate.
On a packet-switching network, packets are still accepted, but delivery delay increases. Thus, if a node has a number of packets queued for transmission, it can transmit the higher- priority packets first. These packets will therefore experience less delay than lower-priority packets. In the virtual circuit approach, a preplanned route is established before any packets are sent.
Once the route is established, all the packets between a pair of communicating parties follow this same route through the network. As a smaller packet size is used, there is a more efficient "pipelining" effect, as shown in Figure However, if the packet size becomes too small, then the transmission is less efficient, as shown in Figure The major differences are that frame relay uses out-of-channel signaling while X.
In frame relay there is no "hop-by-hop" flow control or error control as there is in X. If a frame error is detected it is just dropped rather than being retransmitted. Similarly, on an end-to-end basis, there is no error control or flow control except what is provided by higher level protocols outside of frame relay. On the other hand, because of the lack of hop-by-hop flow control, the user of frame relay has fewer tools to manage network congestion.
The effective use of frame relay also depends on the channels being relatively error free. For example, this is true for fiber optics, but probably not for most forms of broadcast, wireless transmission.
Thus a telephone occupies a circuit for 3 minutes per hour. Each first stage matrix has n input lines and 2n — 1 output lines, so it has n 2n — 1 crosspoints. By the same argument, there are N 2n — 1 crosspoints in the third stage. For large n, we can approximate 2n — 1 by 2n. Circuit Switching vs.
A large noise burst could create an undetected error in the packet. If such an error occurs and alters a destination address field or virtual circuit identifier field, the packet would be misdelivered.
Either can prevent the other from overwhelming it. The layer 3 flow control mechanism regulates the flow over a single virtual circuit. Thus, resources in either the DTE or DCE that are dedicated to a particular virtual circuit can be protected from overflow. Errors are caught at the link level, but this only catches transmission errors. If a packet-switching node fails or corrupts a packet, the packet will not be delivered correctly.
A higher-layer end-to-end protocol, such as TCP, must provide end-to- end reliability, if desired.
Data and Computer Communications (Eighth Edition)
Otherwise, there would have to be global management of numbers. In essence, the upper part of the fraction is the length of the link in bits, and the lower part of the fraction is the length of a frame in bits. So the fraction tells you how many frames can be laid out on the link at one time. Multiplying by 2 gives you the round-trip length of the link. You want your sliding window to accommodate that number of frames so that you can continue to send frames until an acknowledgment is received.
Adding 1 to that total takes care of rounding up to the next whole number of frames. Adding 2 instead of 1 is just an additional margin of safety. See Figure 7. Additionally, error checking is only done on the header in ATM rather than on the whole cell or frame. Virtual channels of ATM that follow the same route through the network are bundled into paths. A similar mechanism is not used in frame relay.
In ATM, virtual channels, which have the same endpoints, can be grouped into virtual paths. All the circuits in virtual paths are switched together; this offers increased efficiency, architectural simplicity, and the ability to offer enhanced network services.
Network transport functions can be separated into those related to an individual logical connection virtual channel and those related to a group of logical connections virtual path. Increased network performance and reliability: The network deals with fewer, aggregated entities. Reduced processing and short connection setup time: Much of the work is done when the virtual path is set up.
By reserving capacity on a virtual path connection in anticipation of later call arrivals, new virtual channel connections can be established by executing simple control functions at the endpoints of the virtual path connection; no call processing is required at transit nodes. Thus, the addition of new virtual channels to an existing virtual path involves minimal processing.
Enhanced network services: The virtual path is used internal to the network but is also visible to the end user. Thus, the user may define closed user groups or closed networks of virtual channel bundles. A user of a VCC is provided with a Quality of Service specified by parameters such as cell loss ratio ratio of cells lost to cells transmitted and cell delay variation.
Switched and semipermanent virtual channel connections: A switched VCC is an on-demand connection, which requires a call control signaling for setup and tearing down.
A semipermanent VCC is one that is of long duration and is set up by configuration or network management action. Cell sequence integrity: The sequence of transmitted cells within a VCC is preserved. Traffic parameter negotiation and usage monitoring: Traffic parameters can be negotiated between a user and the network for each VCC.
The input of cells to the VCC is monitored by the network to ensure that the negotiated parameters are not violated. Virtual channel identifier restriction within a VPC: One or more virtual channel identifiers, or numbers, may not be available to the user of the VPC but may be reserved for network use.
Examples include VCCs used for network management. No framing is imposed. The interface structure consists of a continuous stream of octet cells. Real-time variable bit rate: The principal difference between applications appropriate for rt-VBR and those appropriate for CBR is that rt-VBR applications transmit at a rate that varies with time.. Non-real-time variable bit rate: With this information, the network can allocate resources to provide relatively low delay and minimal cell loss..
Available bit rate: Unspecified bit rate: Guaranteed frame rate: Cell is assigned or on an uncontrolled ATM connection. Cell is unassigned or on an uncontrolled ATM connection. We reason as follows. A total of X octets are to be transmitted. Transmission efficiency N variable 1. For long messages, the optimal achievable efficiency is approached. It is only for very short cells that efficiency is rather low.
However, it does not provide significant gains over fixed-length cells for most values of X. As we have already seen in Problem Packetization Transmission delay ms efficiency 2 1. The transmission time is always incurred so the jitter is due only to the waiting for switches to clear. In the first case the maximum jitter is In the second case the average jitter is Such higher IP-packet loss rate than the cell loss rate is caused by the dropping of cells that are likely to belong to different IP packets.
In order to avoid this high IP-packet loss rate, the Guaranteed Frame Rate GFR service should be used, so that in case of congestion, ATM switches will discard all the cells that comprise a single IP packet, rather than possibly discard one or a few cells from multiple packets.
Because an adaptive routing strategy tends to balance loads, it can delay the onset of severe congestion. There is a tradeoff here between the quality of the information and the amount of overhead. The more information that is exchanged, and the more frequently it is exchanged, the better will be the routing decisions that each node makes.
On the other hand, this information is itself a load on the constituent networks, causing a performance degradation. For each pair of nodes, find a path with the least cost. Dijkstra's algorithm requires that each node must have complete topological information about the network; that is, each node must know the link costs of all links in the network.
The fixed number of hops is 2. The furthest distance from a station is halfway around the loop. Includes PDF lecture slides in Greek. University of Idaho. Includes handouts and many interesting links. University College London. Includes PDF slides.
Queen's U. Includes powerpoint slides. EG Communications Engineering. Includes lecture notes, an number of useful supplement pages. CS Computer Networks and Communications. Includes lecture notes. University of Florida.. EEL Computer Networks. Lots of interesting material. CS Data Communications.
Mount Union College, Alliance, Ohio. Some useful links and interesting concept demonstrations.
CS Data Communication. Includes useful set of PDF and postscript lecture notes. Kasetsart University, Thailand. Includes a very good set of student produced slides and notes in Powerpoint and PDF formats.. At Villanova. Rensselaer at Hartford. If you have any suggestions for site content, please contact me at. In particular, please pass along links to relevant web sites and links to course pages used by instructors teaching from this book.
Network World: Includes keyword-indexed library of RFCs and draft documents as well as many other documents related to the Internet and related protocols. Links to thousands of hardware and software vendors who currently have WWW sites, as well as a list of thousands of computer and networking companies in a Phone Directory.
Good way to keep up on conferences, publications, etc. Has an on-line copy of my article on IPv6, which updates material in the book. Good way to keep up on conferences, publications, etc International Telecommunications Union: International Organization for Standardization: Links to Web pages of vendors, tutorials on technical topics, and other useful information.
Networking Links: OSI History: A brief history of the origins of the OSI model. Wireless Developer Network: News, tutorials, and discussions on wireless topics Siemon Company: Good collection of technical articles on cabling, plus information about cabling standards.. DSL Forum: Forum specifications. The communication task can be organized into three relatively independent layers: A transport B media C physical D processing.
A elastic B real-time C file transfer D transport If the initiator receives confirmation that the requested service has had the desired effect at the other end. Computer games. A discrepancy results if there has been some error in transmission. The receiving TCP performs the same calculation and compares the result with the incoming code. Checksum Secure Shell SSH 7. Flag for inappropriate content.
Jump to Page. Search inside document. Md Saidur Rahman Kohinoor. Ajay Pandey. Sn Lau.
Ashish Singh. Florin Ferchiu. Mehedi Hasan. Student Lecture Notes. Firman Hidayat.FTP provides a basic electronic mail transport facility. Thus UDP provides a useful, though limited, service. Output waveform: Responsible for controlling the operation of the link.
Computer scientists, electrical engineers, and other professionals will find this up-to-date, accessible volume an essential resource about key areas, problems, and methods associated with data coding.
University of Idaho. With this information, the network can allocate resources to provide relatively low delay and minimal cell loss..
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