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The instructions and questions for the third electronic and computer engineering examination for the communication systems engineering module (ee314) during semester ii, 2008/2009. Details about the exam code, duration, required materials, and specific questions related to huffman coding, convolutional encoding, synchronous and asynchronous transmission systems, ip packet fragmentation, and routing.
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Exam Code(s) 3BN 3BP Exam(s) Third Electronic Engineering Examination Third Electronic and Computer Engineering Examination Module Code(s) EE Module(s) Communication Systems Engineering Paper No. 1 Repeat Paper No External Examiner(s) Prof. G.W. Irwin Internal Examiner(s) Prof. G. Ó Laighin Mr. L. Kilmartin Instructions: Answer any five questions from seven. All questions carry equal marks (20 marks each) Duration 3 hrs No. of Pages 5 Department(s) Electronic Engineering Course Co-ordinator(s) Mr. L. Kilmartin Requirements : MCQ Handout Statistical Tables Graph Paper Log Graph Paper Other Material Standard mathematical tables
1. (a) A source coding subsystem must be designed to allow the transmission of data from a remote monitoring unit. This data, which indicates the state of the monitor, is represented by means of a symbol value. It is decided to examine the data being produced by the sensor before the source coding module is designed. Over a 24 hour period the number of occurrences of each of the six possible symbols in the data generated by the sensor was recorded by an automatic logging system. The results of this log were: Symbol S 1 S 2 S 3 S 4 S 5 S 6 Number of Occurrences 30 33 5 20 52 10 You are required to design, AND determine the efficiency of, a Huffman source coding scheme for this system based on the information presented above. [7 marks] (b) Figure 1 describes the structure of a convolutional encoder formed by a three-bit shift register which generates two output bits for every input bit. Firstly, draw state transition and trellis diagrams for this particular encoder structure. The receiver for this convolutional coder receives the following bit sequence: 10 01 00 01 11 Determine the most probable binary input sequence which was applied to the transmitter’s convolutional encoder. You may assume that this sequence was generated using the standard initial and final state constraints for a convolutional coder of this type. SSS 111 S 2 S 3 Xin V 1 V 2 Xout XOR Figure 1 - Structure of Convolutional Encoder [9 marks] (c) Interleaving is a process which is often used as a mechanism for alleviating the effects of burst errors. Explain what you understand by the two terms in bold using an interleaved Hamming coding scheme as an example. [4 marks] 2. (a) Clearly distinguish between a synchronous and an asynchronous transmission system and hence explain how line coding/waveform formatting techniques are used to achieve bit or clock synchronisation. Outline the difference between the function of bit synchronisation and (a) byte synchronisation and (b) block synchronisation in such transmission systems. [6 marks] (b) Table 1 contains the sequence of waveforms which are received at a HDB3 receiver circuit. The symbols +, - and 0 represent a positive polarity pulse, negative polarity pulse and zero volts, respectively. Determine the binary values which these symbols represent and clearly indicate any violation/protection pulses present in the sequence. You may assume that the last pulse received prior to this sequence was of negative polarity.
Distinguish clearly between connection oriented and connectionless services implemented on PSDNs and identify ONE relative advantage and ONE relative disadvantage which the latter form of service has over the former type of service. [10 marks] (b) Explain what is meant by the term Maximum Transmission Unit (MTU) and outline how this value affects the process of IP packet fragmentation. Your answer should clearly identify and explain all fields in the standard IP header which are used during packet fragmentation. An internetwork is formed by two different networks as outlined below: Network A has a 4000 byte MTU Network B has a 2000 byte MTU A 5000 byte frame is transmitted by an IP host on Network A to another IP host on Network B. Determine how this PDU is fragmented on each network. Clearly indicate the values used in the relevant parameters in the IP header of each fragment. [5 marks] (c) An organisation requires in the region of 8000 IP nodes. Determine the address utilisation factor if the network is allocated a (i) single class B address and (ii) a suitable group of Class C addresses. Hence, explain how the concept of classless addressing could be used if the organisation was allocated specific class C addresses and give ONE advantage and ONE disadvantage of this approach compared to the traditional class based addressing schemes. [5 marks]
4. (a) An IP packet arrives at a router to which the destination node of the packet is directly connected via a broadcast network topology. Explain why it is necessary for the router to perform an address translation from the destination IP address using the Address Resolution Protocol (ARP). [5 marks] (b) Contrast the relative advantages and disadvantages of using a static IP address allocation procedure versus a dynamic IP address allocation procedure based on the Dynamic Host Configuration Protocol (DHCP) [ 5 marks] (c) Clearly explain your understanding of the term routing and hence outline the difference between the aims and operations of inter-domain and intra-domain routing protocols in an IP network. [5 marks] (d) Determine the routing table at node C in the network of Figure 3 below using the forward search algorithm. Your answer MUST show the steps used to determine the final table.
Figure 3 – Network Structure [5 marks]
5. (a) Comment on the support offered by (i) UDP and (ii) TCP for each of the following functionalities: (i) Transport Layer Addressing (ii) Reliability (iii) Flow Control (iv) Congestion Control By citing an example of the type of application that would utilise them, justify the need for both UDP and TCP transport layer protocols. [6 marks] (b) Explain why it is necessary to have an adaptive timeout duration for the TCP sliding window protocol and clearly identify the impact of utilising (i) too short of a timeout duration or (ii) too long of a timeout duration. A TCP implementation measures the round trip time (RTT) for a number of specific TCP segment as 125 ms, 140 ms and 110 ms, in that order. The estimated RTT prior to the first of these particular TCP segments being transmitted is 120 ms. Determine the TCP timeout duration at the end of this period of time using: (i) The original TCP timeout update algorithm (using =0.9). (ii) The Jacobson-Karels algorithm (with =1, =4, =0.1 and an initial deviation of 10 ms). If the real RTT process is actually normally distributed with a mean of 125 ms and a standard deviation of 20 ms, estimate the percentage of unnecessary re-transmissions which will take place based on the timeout value determined using (ii) above. [9 marks] (c) Clearly distinguish between the role and operation of the flow control and congestion control algorithms as implemented in standard TCP. [5 marks]