Telecommunications and Networking - Reliability and Quality of Service - Assignment Example

Student name Course name Lecturer Name Question 1 The overall effectiveness of a communication network or channel can be measured in terms of its Quality of Service requirements and specifications. Modern businesses are in pursuit of higher speed communication systems that are able to save time and make businesses run smoothly. Some of the properties that are used in deciding the appropriate communication applications or network system are their reliability, jitter, delay and bandwidth requirements. These terms have specific meaning within the Quality of Service domain. a) Reliability Reliability is an important nun-functional requirement of any communication system. To start with, the reliability of a communication application refers to the measure of how much it is available to be used. The availability here includes the ability to work within acceptable standards. A communication that is available to be used but running at slower speeds is not said to be reliable. Mean Time between Failures (MTBF) is a measure of the average time a component is expected to operate between failures. This is always provided by the manufacturer and is useful in the purchase of such systems. A system that is reliable experiences no, or very minimal failures. Such failures could result from hardware problems. Software malfunction and can also result from the data carrying medium and the network operating system. Junction networking, web Quality of Service publication indicates that reliability is one of the policies provided by a Data Distribution Service (DDS) and it dictates whether the DDS is performing in the required manner. This means at all costs, to maintain the reliability of the communication system, precautions have to be made to minimize all kinds of possible system failures, so as to have a higher Mean Time Between failures rating. The reliability of a system is known by performing reliability tests. The reliability tests as opposed to other system specification tests are not binary, that is, it is not easy to tell if a system meets or does not meets its requirements. This is because there are different dependencies according to the end user of that communication system, and the view by a special analyst or tester. Reliability can be versioned in two aspects, hardware and software reliability. No doubt any hardware on a communication system is prone to wear and failure, hence this is measured in terms of time to failure or time between failures. Software reliability on the other hand can be assessed on different aspects by testing reaction to different ranges of communication inputs. Generally, the reliability of a communication system can be seen as the probability that a system will operate without failure for a number of uses, over a period of time, says M Murphy(2007) as he addresses the Reliability requirements of a communication system. b) Delay (Latency) In communication technology, latency or delay is the time from the source sending a packet of data to the time the destination receives it. Packet loss and jitter are some other factors that affect the delay in a communication system. In arguably all systems, latency consist of three elements namely codec delay, playout delay and network delay all of which contribute to the cumulative delay of the system. Codec delay is the delay in coding and decoding instruments that act data streams incident to them. Network delay on the other hand is basically the media of packet data transfer. Delays are caused by distance, software and hardware errors, recovery from those errors, the processing capabilities of systems involved in the transmission, congestion of the communication channel being engaged and exterior physical factors such as barriers and interference. Removing some hardware delays still leaves a number of software delays available. (Huber and Robert 2013, p.252). Delays due to distance normally called propagation delays are prominent in cross country data transmission. This delay not only results from the effects of the media used, but also the availability of numerous devices along the way that add up to this delay. Latency can be a bit of a problem when it comes to audio system. Taking an example of the Microsoft Windows Drivers which have a latency of about 500ms. It is only possible to reduce this by supported interface optimization to scales that the human ear cannot perceive. In satellite communication, latency rises at the top above other factors like bit error rate, availability, reliability, circuit speeds, geographic coverage or cost-effectiveness. Another view of latency in satellite communication apart from the propagation delay is the processing delay, which the cumulative delay caused by ever software and hardware that a packet of information passes through. Quality of Service requirements stipulates that a communication system or channel should have as minimal latency as possible. c) Jitter. The deviation of a noisy signal from its ideal can be viewed from two aspects, timing deviation and amplitude deviation. The amplitude deviation is also called noise but the deviation of time (Δt) is defined as the timing jitter (or simply jitter). Jitter can also be defined as the variation of the packet latency of a communication system. There is no doubt that jitter impacts negatively to any communications system. This impact of jitter can be examined from the perspective of a receiver for a communication system. The presence of a jitter causes a system Bit Error Rate (BER), therefore compromising the Quality of Service of a communications System. Since bit error rate is the bottom line in the determination of a good communication system, the presence of jitter is by no chance allowed in the Quality of Service specifications and is recommended to be as low as possible, which is however only available in ideal instruments. Data transmission is critical and thus the management of jitter is an important activity in data transmission. There are three types of jitter namely: i) Random jitter – this results from related clock timing or unpredictable electronic timing noise. Random jitter is also known as unbounded jitter. ii) Deterministic jitter – this is the type of jitter that can be predicted and determined. iii) Total jitter – this value is calculated from the beat error rate, as well as the combined involvement of the random and deterministic jitters. Since the management of jitter is vital in successful communication of video, data and computerized signals, jitter mitigation techniques are implemented. They include: Jitter buffer – this is used to eliminate jitter in video and audio systems. Anti-jitter circuits: These consist of a group of electronic circuits and Dejitterizer: This is an elastic buffer in which a signal is temporarily stored and transmitted at the average incoming signal rate. This method is however not effective in mitigation in communication of low frequency. (Janssen 2009, Definition). d) Bandwidth(Throughput) In a broader telecommunication perspective bandwidth can be classified into network bandwidth, data bandwidth, or digital bandwidth. All of them can be as well taken as bandwidth and is defined as a measurement of data communication resources available or that is used in units of bit-rates usually in bits per second of higher multiples of it. In other disciplines and topics such as signal processing and wireless communication, bandwidth refers to the analog signal bandwidth that is measured in Hertz. This is normally the difference between the highest and the lowest available frequency in an analog frequency system like in frequency Modulation. Bandwidth, the data bandwidth concept, is the one that is in relation to the Quality of service in the data communication. Light applications used in email sending over the internet make use of fewer data communication resources per unit time while more robust applications like those used in video conferencing and webinars have maximum bandwidth usage. Businesses and companies would like to reduced the physical distance between managers by virtually having board meetings via video conferencing, thus, larger companies have a large requirement of bandwidth but at a cheaper cost. Start ups on the other hand need increasingly large bandwidth since they are just on the brink of starting to use hosted information systems that are not located on their premises. With the clogging of the many users demanding for large chunks of bandwidth from information systems,, consistency and reliability is difficult to maintain this compromising some of the Quality of Service protocols. The bandwidth of a data network communication influences the propagation and processing speeds making businesses to opt for larger bandwidth system. Question 2 In a bid to satisfy the expectation of customers in the communication industry, a lot is being done that the Quality of Service provided by these communication networks, media and applications meet certain criteria that upholds the integrity of the communication system in order to remain ace[table by the end user. The foregoing discussion sheds light on some of the techniques that are used to improve the Quality of Service. a) Scheduling. In a communication system, packets from various flows arrive at the processing unit which may be a switch or a router from various sources. These packets carry different kinds, sizes and level of information and with different priority requirements. A good scheduling technique to treat the flows in a fair and appropriate manner is in demand. There are also various scheduling techniques that are implemented. i) FIFO Queueing. The acronym stands for First In First Out. In this technique, the flows are stored in a First Come First Serve basis, so that the other priority requirements of the different flows are not considered but the time at which they arrived at the processing unit. The flows are processed one by one and the processing of the next only begins after one has been processed. When the processing speed is more than the packets arrival rate, the queue is faster emptied and more room is left for other flows to get in for processing. Trouble gets in when the processing speed is less than the arrival rate, in which case the queue is filled up and any extra packets coming in are discarded. FIFO scheduling is the default scheduling technique in most systems and in most cases do not need configuration. One advantage of FIFO scheduling is that it takes care of network congestion and allows earlier arrivals to be processed first. This ensures that the new arrivals are kept waiting when there is congestion and released for processing when there is no congestion. However, a major shortcoming of FIFO is that it takes no consideration of packet priority, which is in some cases most important. Moreover, it doe not provide any protection mechanism against malicious applications. Better scheduling algorithms have since been applied to solve the shortcomings of the FIFO technique. ii) Priority Queuing. In priority queuing, the packets from different flows are first assigned priority classes immediately after arrival at the switch board. Thos packets with same priority are placed in a queue and the ones with the highest priority are processed first ant the process continues until the queues are finished. Priority queuing is flexible as to prioritize according to network protocol, incoming interface and even network size. The algorithm gives the higher priority packets preferential treatment better than the lower priority packets. b) Traffic Shaping This is a technique to control the amount and rate of traffic being sent to a network. There are techniques used to do this. i) Leaky bucket. Here, when the traffic consists of fixed size packets, the process removes a fixed number of packets from the queues while when the traffic consists of variable length packets, the fixed output rate must be based on the number of bytes or bits. In page 56 of his book, Behrouz (2003) argues that leaky bucket is used to check that data transmissions, in the form of packets, conform to defined limits on bandwidth and burstiness (a measure of the unevenness or variations in the traffic flow) . ii) Token bucket. Leaky bucket algorithm outputs the data in average rate from the burst data, but it does not taken the time when the host was idle, into account. The token bucket reduces the out band traffic flow to evade congestion by constraining specific traffic to a specified bit rate. This has the implication that traffic that is adhering to certain profile can be shaped to meet downstream requirements. c) Admission control. From its name, this is an algorithm used by a switch or a router to admin or rejects an incoming flow based on some predetermined parameters that are called flow specifications. Before a flow is forwarded for processing, the router checks the flow specification. This is done to evaluate if its capacity and its previous commitments to other flows can allow the admission of other flows. In other words, admission control is set of online actions that must be taken during the flow stablishment (invocation) phase to determine whether the flow should be admitted or not. In order for an admission control algorithm to succeed it is necessary that it fulfils some requirements: Robustness – It must ensure that the desired Quality of Service provided and that it is robust with respect to the traffic hierarchy and topology. Resource utilization.- one of the objectives of admission control is to maximize resource usage. Cost of implementation – From business stand point, the cost of employing an admission control algorithm must be smaller than its benefits. Similarly, the traffic properties required must be easily obtained. These specifications bring us to a conclusion that the challenge existing is to develop a simple admission control algorithm that provide the desired Quality of Service and improve network efficiency in any possible traffic conditions. (Shiomoto et al 1999, p.98). d) Resource reservation The flow of data in any communication system needs resources such us CPU time and bandwidth. For these resources to be available at the time they are needed, they must be reserved beforehand. This algorithm ensures that before any operation or data transmission, the resources necessary to be utilized during the process are prepared to work. Resource reservation ins initiated by the Resource reservation protocol (RSVP) that is a transport layer protocol operating on IPV4 and IPV6 which provided receiver initiated setup for resource reservations. RSVP is used by either hosts or routers to request and/or deliver certain levels of quality of service for application data streams or flows. This protocol defines how applications place reservations and how they hand them over once their usage is ended. When each of the nodes releases the resources after their operation, there is a cumulative effect of resources at each node of the network that helps in the improvement of the Quality of Service of the communication. The moment a router receives a resource reservation message, it will handle that message in the following ways: Make a corresponding reservation based on the request parameters. The admission control and the policy controls process the parameters but if they do not support the reservation being made, they send an error message. Forward the upstream request. At this point, the router stores the nature of the store and patrols it. With the foregoing explanation, resource reservation in the end increases the Quality of service of a communication application or system. In conclusion, it is useful to note that, modern businesses will never stop at anything to make communication easier and faster so as to minimize time loss and increase business opportunities. This has in the meantime created an extra demand for telecommunication resources. If all are going to demand very large shares, than there will be a lot of congestion in the communication systems. On my opinion, a possible solution to maintaining a good communication system is the use of reliable shares and enhances Quality of Services using the various available techniques. Bibliography. Junction Networks ,2012“Reliability and Quality of service” http://www.junctionnetworks.com/knowledgebase/onsip/faqs/reliability-and-quality-of-service . Retrieved July 31st 2014. M. Murphy (2007 Oct) “Reliability requirements.” http://requirements.seilevel.com/blog/2007/10/reliability-requirements.html Retrieved July 31 2014. Huber, David M., and Robert E. Runstein. "Latency." Modern Recording Techniques. 7th ed. New York and London: Focal, 2013. 252. Print. Cory Janssen 2009 “Jitter: Definition” Techopedia. http://www.techopedia.com/definition/3041/jitter-voip-voice-over-internet-protocol-voip Retrieved 31st July 2014. Behrouz A. Forouzan 2007 Data communications and networking, McGraw-Hill.p.56 Andrew S. Tanenbaum, 2003 Computer Networks, Fourth Edition, ISBN 0-13-166836-6, Prentice Hall PTR ITU-T, Traffic control and congestion control in B ISDN, Recommendation I.371, International Telecommunication Union, 2004, Annex A, page 87. K. Shiomoto, N. Yamanaka and T. Takahashi , Overview of Measurement-based Connection Admission Control Methods in ATM Networks, IEEE Communication Surveys 1999. p.98 Read More