Sunday, October 6, 2019
Daily Social Responsibility in Action Assignment
Daily Social Responsibility in Action - Assignment Example The involved actions in social responsibility may not necessarily remedy the actual effects on the society but aim achieving an overall social benefit. Organizations also engage in social responsibility in order to meet interest of people who influence the organizationââ¬â¢s existence and operations. Partnerships and other contracts may for example be tied to terms for corporate social responsibility and this forces organization to honor their obligations to the stakeholders (Grossling, 2011). Another reason for organizationââ¬â¢s participation in social responsibility is that the role averts pressure that would have built against the organizationââ¬â¢s interest such as crimes into vandalism of assets (Visser, Matten, Pohl and Tolhurst, 2010). Organizations derive diversified benefits from their participation in social responsibility. Such benefits include reduced costs due to recycled resources, innovation into new products, motivated staff and a safer environment. The organizationââ¬â¢s employees however benefit from ââ¬Å"environmental awareness,â⬠and ââ¬Å"improved staff moraleâ⬠towards higher output levels (Frank and Neergaard, 2012, p. 86). My experience with a company that involved in corporate social responsibility involves witnessing academic sponsorship by a commercial bank to needy students. The bankââ¬â¢s partners initiated this role and it improved customerââ¬â¢s loyalty to the bank, especially people from the beneficiary
Saturday, October 5, 2019
Perceptions that consumers have for the different marketing messages Essay
Perceptions that consumers have for the different marketing messages - Essay Example This research will begin with the statement that the selective exposure process focuses on individuals to agree with those medium of communication that is in alignment with their views and opinions. Consumers in this selective process only go out for things that are of interest to them and oppose those things that they are against about. An example can be the fluctuations of the share prices. A drop in the share prices would affect the consumers and they may apply the selective exposure process. The selective retention process observes that consumers tend to retain those marketing messages that are of interest to them and also are favoring their opinions as well. The marketing messages that are against the opinion of the consumers do not tend to retain for long in the minds of consumers. The products advertised to consumers such as mobile phones for youngsters may be a highly attractive product and they may observe keenly the advertising done for mobile phones. The selective percepti on theory states that consumers interpret facts that they are interested in. In other words, consumers comprehend the situation the way they want to see it as. They hear what they believe in rather than what the message is actually trying to state. Therefore, in this case for different consumers, the same message may have different worth and meaning to them. For example, consumers that like to watch a lot of television may only see the advantageous side of watching television and may ignore the disadvantages that watching too much television has on humans.
Friday, October 4, 2019
Poetry Slam Movie Review Example | Topics and Well Written Essays - 500 words
Poetry Slam - Movie Review Example "Slam's victory at Sundance marks a critical move for the art. Poetry's value is completely tied to its integrity, and in Slam it is the poem that defines the terms. Saul Williams and Sonja Sohn, the star-crossed fiery loves at the centre of this story, are both seasoned poets in the New York spoken-word world The poets in Slam speak a poem through film, they do not sit on a plastic waiting for the pan across a furled eyebrow." (Stratton and Wozencraft, 137) Therefore, it is essential to comprehend that the film Slam celebrates its success through the depiction of the relevance and impact of poetry in the contemporary society and the major characters, themes, action, etc centre around the impact of poetry in society. The relevance and impact of poetry in the contemporary society has been the central idea suggested by the film Slam and the film has been central in representing the great role of poetry, rap, performance art and stand-up comedy etc in our society. It is a movie which specifically suggests the relevance of poetry in human life and society. "Slam is a raw poem of a movie.
Thursday, October 3, 2019
Poverty Policy In The Land Of Milk And Honey Essay Example for Free
Poverty Policy In The Land Of Milk And Honey Essay People might think that poverty is the last thing to be occurred in the United States of America ââ¬âthe land of milk and honey as they say, because America is known for having a Tiger Economy ever since. But it is a shame should the government of America admits that they too can experience poverty? Let them allow having an excuse that nothing is perfect. Everything is possible. However, we all know that we can attain the 0% rate of poverty in our country if everyone will work it out together. How the poverty is being measured in the United States of America? The United States Department of Health and Human Services says that there are two slightly different versions of the federal poverty measure. One is the poverty threshold which is used mainly in Census Bureau for statistical purposes, and the other one is the poverty guidelines which are basically for administrative purposes. (2007). But the US government do not really understand that the only thing this policy or measurement can do is the knowledge about figures and recoded data of how many could pass yearly in the poverty line in accordance to the standards that they have set. They never realize that the poverty measurement has nothing to do with poverty itself and how to totally demolish the poverty problem in the United States of America ââ¬â the land of milk and honey! As a citizen, all I can say is that we have to go back to the main roots of the problem. What are those problems that lead us to poverty? Are we ready to face them? I guess we should. And it is not just facing them but to act on them for the betterment of any individual or household concern but also for the good of the common people, government and country. Poverty has not to be measured in starvation and emptiness only. There are many factors involve including illiteracy, illegitimacy, immorality, unemployment, dirty politics, bureaucracy, environment, lifestyle, vices, crime, over population, sickness, mortality or even having your own identity and citizenship. But as long as the scarcity and level of consumptions of every household are the major factors that we keep on measuring, we will never resolve the problem of poverty. Again, I suggest that we go back to the main roots of the problem. I affirm to David Brooksââ¬â¢ optimism outlook in his argument ââ¬Å"â⬠¦these rapid improvements (which refers to globalization) at the bottom of the income ladder are contributing to and correlating with declines in illiteracy, child labor rates and fertility rate. â⬠(2004). But there are also some things that should be remembered. Tiger economy or third world country has the equal opportunity of experiencing poverty. However, addressing the issue of poverty lies not to the government alone. Every human beings living on Earth has to do their fair share of opinions and actions in fighting poverty. It could be done by helping others or the nation. But I guess the best poverty alleviation policy is improving first your own quality of life wherever you are; hence, every place you could have been has the chance to be a poverty-free land of milk and honey.
Wednesday, October 2, 2019
Analysis of QoS Parameters
Analysis of QoS Parameters Chapter 3 3. Analysis of QoS Parameters 3.1 Introduction A Number of QoS [11] of parameters can be measured and monitored to determine whether a service level offered or received is being achieved. These parameters consist of the following 1. Network availability 2. Bandwidth 3. Delay 4. Jitter 5. Loss 3.1.1 Network Availability Network availability can have a consequential effect on QoS. Simply put, if the network is not available, even during short periods of time, the user or application may achieve unpredictable or undesirable performance (QoS) [11]. Network availability is the summation of the availability of many items that are used to create a network. These include network device redundancy, e.g. redundant interfaces, processor cards or power supplies in routers and switches, resilient networking protocols, multiple physical connections, e.g. fiber or copper, backup power sources etc. Network operators can increase their networks availability by implementing varying degrees of each item. 3.1.2 Bandwidth Bandwidth is one of the most important QoS parameter. It can be divided in to two types 1. Guaranteed bandwidth 2. Available bandwidth 3.1.2.1 Guaranteed bandwidth Network operators offer a service that provides minimum BW and burst BW in the SLA. Because the guaranteed BW the service costs higher as compare to the available BW service. So the service providers must ensure the special treatment to the subscribers who have got the guaranteed BW service. The network operator separates the subscribers by different physical or logical networks in some cases, e.g., VLANs, Virtual Circuits, etc. In some cases, the guaranteed BW service traffic may share the same network infrastructure with available BW service traffic. We often use to see the case at location where network connections are expensive or the bandwidth is leased from another service provider. When subscribers share the same network infrastructure, the subscribers of the guaranteed BW service must get the priority over the available BW subscribers traffic so that in times of networks congestion the guaranteed BW subscribers SLAs are met. Burst BW can be specified in terms of amount and du ration of excess BW (burst) above the guaranteed minimum. QoS mechanism may be activated to avoid or discard traffic that use consistently above the guaranteed minimum BW that the subscriber agreed to in the SLA. 3.1.2.2 Available bandwidth As we know network operators have fixed Bandwidth, but to get more return on the investment of their network infrastructure, they oversubscribe the BW. By oversubscribing the BW a user is subscribed to be no always available to them. This allows users to compete for available BW. They get more or less BW it depends upon the amount of traffic form other users on the network at any given time. Available bandwidth is a technique commonly used over consumer ADSL networks, e.g., a customer signs up for a 384-kbps service that provides no QoS (BW) guarantee in the SLA. The SLA points out that the 384-kbps is standard but does not make any guarantees. Under lightly loaded conditions, the 384-kbps BW will be available to the users but upon network loaded condition, this BW will not be available consistently. It can be noticed during certain times of the day when number of users access the network. 3.1.3 Delay Network delay is the transit time an application experiences from the ingress (entering) point to the egress (exit) point of the network. Delay can cause significant QoS issues with application such as Video conferencing and fax transmission that simply time-out and final under excessive delay conditions. Some applications can compensate for small amounts of delay but once a certain amount is exceeded, the QoS becomes compromised. For example some networking equipment can spoof an SNA session on a host by providing local acknowledgements when the network delay would cause the SNA session to time out. Similarly, VoIP gateways and phones provide some local buffering to compensate for network delay. There can be both fixed and variable delays. Examples of fixed delays are: Application based delay, e.g., voice codec processing time and IP packet creation time by the TCP/IP software stack Data transmission (queuing delay) over the physical network media at each network hop. Propagation delay across the network based on transmission distance Examples of variable delays are: â⬠¢ Ingress queuing delay for traffic entering a network node â⬠¢ Contention with other traffic at each network node â⬠¢ Egress queuing delay for traffic exiting a network node 3.1.4 Jitter (Delay Variation) Jitter is the difference in delay presented by different packets that are part of the same traffic flow. High frequency delay variation is known as jitter and the low frequency delay variation is known as wander. Primary cause of jitter is basically the differences in queue wait times for consecutive packets in a flow and this is the most significant issue for QoS. Traffic types especially real time traffic such as video conferencing can not tolerate jitter. Differences in packet arrival times cause in the voice. All transport system exhibit some jitter. As long as jitter limits below the defined tolerance level, it does not affect service quality. 3.1.5 Loss Loss either bit errors or packet drops has a significant impact on VoIP services as compare to the data services. During the transmission of the voice, loss of multiple packets may cause an audible pop that will become irritating to the user. Now as compare to the voice transmission, in data transmission loss of single bit or multiple packets of information will not effect the whole communication and is almost never noticed by users. In case of real time video conferencing, consecutive packet loss may cause a momentary glitch (defect) on the screen, but the video then proceeds as before. However, if packet drops get increase, then the quality of the transmission degrades. For minimum quality rate of packet loss must be less than 5% and less then 1% for toll quality. When the network node will be congested, it will drop the packets and by this the loss will occur. TCP (Transmission Control Protocol) is one of the networking protocols that offer packets loss protection by the retransmission of packets that may have been dropped by the network. When network congestion will be increased, more packets will be dropped and hence there will be more TCP transmission. If congestion continues the network performance will obviously degrade because much of the BW is being used for the retransmission of dropped packets. TCP will eventually reduce its transmission window size, due to this reduction in window size smaller packets will be transmitted; this will eventually reduce congestion, resulting in fewer packets being dropped. Because congestion has a direct influence on packet loss, congestion avoidance mechanism is often deployed. One such mechanism is called Random Early Discard (RED). RED algorithms randomly and intentionally drop packets once the traff ic reaches one or more configured threshold. RED provides more efficient congestion management for TCP-based flows. 3.1.5.1 Emission priorities It determines the order in which traffic is transmitted as it exits a network node. Traffic with higher emission priority is transmitted a head of traffic with a lower emission priority. Emission priorities also determine the amount of latency introduced to the traffic by the network nodes queuing mechanism. For example, email which is a delay tolerant application will get the lower emission priority as compare to the delay sensitive real time applications such as voice or video. These delay sensitive applications can not be buffered but are being transmitted while the delay tolerant applications may be buffered. In a simple way we can say that emission priorities use a simple transmit priority scheme whereby higher emission priority traffic is always transmitted ahead of lower emission priority traffic. This is typically accomplished using strict priority scheduling (queuing) the downside of this approach is that low emission priority queues may never get services (starved) it there is always higher emission priority traffic with no BW rate limiting. A more detailed scheme provides a weighted scheduling approach to the transmission of the traffic to improve fairness, i.e., the lower emission priority traffic is transmitted. Finally, some emission priority schemes provide a mixture of both priority and weighted schedulers. 3.1.5.2 Discarded priorities Are used to determine the order in which traffic gets discarded. Due to the network congestion packets may be get dropped i.e., the traffic exceeds its prescribed amount of BW for some period of time. When the network will be congested, traffic with a higher discard priority will get drop as compare to the traffic with a lower discard priority. Traffic with similar QoS performance can be sub divided using discard priorities. This allows the traffic to receive the same performance when the network node is not congested. However, when the network node gets congested, the discard priority is used to drop the more suitable traffic first. Discard priorities also allow traffic with the same emission priority to be discarded when the traffic is out of profile. With out discard priorities traffic would need to be separated into different queues in a network node to provide service differentiation. This can be expensive since only a limited number of hardware queues (typically eight or less) are available on networking devices. Some devices may have software based queues but as these are increasingly used, network node performance is typically reduced. With discard priorities, traffic can be placed in the same queue but in effect the queue is sub divided into virtual queues, each with a different discard priority. For example if a product supports three discard priorities, then one hardware queues in effect provides three QoS Levels. Performance Dimension Application Bandwidth Sensitivity to Delay Jitter Loss VoIP Low High High Medium Video Conf High High High Medium Streaming Video on Demand High Medium Medium Medium Streaming Audio Low Medium Medium Medium Client Server Transaction Medium Medium Low High Email Low Low Low High File Transfer Medium Low Low High Table 3.1: Application performance dimensions (use histogram) Table 3.1 illustrates the QoS performance dimensions required by some common applications. Applications can have very different QoS requirements. As these are mixed over a common IP transport network, without applying QoS the network traffic will experience unpredictable behavior. 3.2 Categorizing Applications Networked applications can be categorized based on end user application requirements. Some applications are between people while other applications are a person and a networked device application, e.g., a PC and web server. Finally, some networking devices, e.g., router-to-router. Table 3.2 categorizes applications into four different traffic categories: 1. Network Control 2. Responsive 3. Interactive 4. Timely Traffic Category Example Application Network Control Critical Alarm, routing, billing ETC. Responsive Streaming Audio/Video, Client/Server Transaction Interactive VoIP, Interactive gaming, Video Conferencing Timely Email, Non Critical Table 3.2: Application Categorization 3.2.1 Network Control Applications Some applications are used to control the operations and administration of the network. Such application include network routing protocols, billing applications and QoS monitoring and measuring for SLAs. These applications can be subdivided into those required for critical and standard network operating conditions. To create high availability networks, network control applications require priority over end user applications because if the network is not operating properly, end user application performance will suffer. 3.2.2 Responsive applications Some applications are between a person and networked devices applications to be responsive so a quick response back to the sender (source) is required when the request is being sent to the networking device. Sometimes these applications are referred to as being near real time. These near real time applications require relatively low packet delay, jitter and loss. However QoS requirements for the responsive applications are not as stringent as real time, interactive application requirements. This category includes streaming media and client server web based applications. Streaming media application includes Internet radio and audio / video broadcasts (news, training, education and motion pictures). Streaming applications e.g. videos require the network to be responsive when they are initiated so the user doesnt wait for long time before the media begins playing. For certain types of signaling these applications require the network to be responsive also. For example with movie on deman d when a user changes channels or forward, rewinds or pause the media user expects the application to react similarly to the response time of there remote control. The Client / server web applications typically involve the user selecting a hyperlink to jump from one page to another or submit a request etc. These applications also require the network to be responsive such that once the hyperlink to be responsive such that once the hyperlink is selected, a response. This can be achieved over a best effort network with the help of broadband internet connection as compare to dial up. Financial transaction may be included in these types of application, e.g., place credit card order and quickly provide feedback to the user indicating that either the transaction has completed or not. Otherwise the user may be unsure to initiate a duplicate order. Alternatively the user may assume that the order was placed correctly but it may not have. In either case the user will not be satisfied with the network or applications performance. Responsive applications can use either UDP or TCP based transport. Streaming media applications typically use UDP because in UDP it would not be fruitful to retransmit the data. Web based applications are based on the hypertext transport protocol and always use TCP, for web based application packet loss can be managed by transmission control protocol (TCP) which retransmit lost packets. In case of retransmission of lost streaming media is sufficiently buffered. If not then the lost packets are discarded. This results in the form of distortion in media. 3.2.3 Interactive Applications Some applications are interactive whereby two or more people communicate or participate actively. The participants expect the real time response from the networked applications. In this context real time means that there is minimal delay (latency) and delay variations (jitter) between the sender and receiver. Some interactive applications, such as a telephone call, have operated in real time over the telephone companies circuit switched networks for over 100 years. The QoS expectations for voice applications have been set and therefore must also be achieved for packetized voice such as VoIP. Other interactive applications include video conferencing and interactive gaming. Since the interactive applications operate in real time, packet loss must be minimized. Interactive applications typically are UDP based (Universal Datagram Protocol) and hence cannot retransmit lost or dropped packets as with TCP based applications. However it would not be beneficial to retransmit the packets because interactive applications are time based. For example if a voice packet was lost. It doesnt make sense to retransmit the packet because the conservations between the sender and receiver have already progressed and the lost packet might be from part of the conversation that has already passed in time. 3.2.4 Timely Applications There are some applications which do not require real time performance between a person and networked devices application but do require the information to be delivered in a timely manner. Such example includes save and send or forward email applications and file transfer. The relative importance of these applications is based on their business priorities. These applications require that packets arrive with abounded amount of delay. For example, if an email takes few minutes to arrive at its destination, this is acceptable. However if we consider it in a business environment, if an email takes 10 minutes to arrive at its destination, this will often not acceptable. The same bounded delay applies to file transfer. Once a file transfer is initiated, delay and jitter are illogical because file transfer often take minutes to complete. It is important to note that timely applications use TCP based transport instead of UDP based transport and therefore packet loss is managed by TCP which r etransmit any lost packets resulting in no packet loss. By summarizing above paragraph we can say that timely applications expect the network QoS to provide packets with a bounded amount of delay not more than that. Jitter has a negligible effect on these types of applications. Loss is reduced to zero due to TCPs retransmission mechanism. 3.3 QoS Management Architecture We can divide QoS management architecture of VoIP into two planes: data plane and control plane. Packet classification, shaping, policing, buffer management, scheduling, loss recovery, and error concealment are involved in the mechanism of data plane. They implement the actions the network needs to take on user packets, in order to enforce different class services. Mechanisms which come in control plane are resource provisioning, traffic engineering, admission control, resource reservation and connection management etc. 3.3.1 Data Plane 3.3.1.1 Packet Forwarding It consists of Classifier, Marker, Meter, Shaper / Dropper. When a packet is received, a packet classifier is used to determine which flow or class the packet belongs to. Those packets belong to the same flow/class obey a predefined rule and are processed in an alike manner. The basic criteria of classification for VoIP applications could be IP address, TCP/UDP port, IP precedence, protocol, input port, DiffServ code points (DSCP), or Ethernet 802.1p class of service (CoS). Cisco supports several additional criteria such as access list and traffic profile. The purpose of the meter is to decide whether the packet is in traffic profile or not. The Shaper/Dropper drops the packets which crossed the limits of traffic profile to bring in conformance to current network load. A marker is used to mark the certain field in the packet, such as DS field, to label the packet type for differential treatment later. After the traffic conditioner, buffer is used for packet storage that waits for transmission. 3.3.1.2 Buffer Management and Scheduling Active queue management (RED) drops packets before the repletion of the queue can avoid the problem of unfair resource usage. Predictable queuing delay and bandwidth sharing can be achieved by putting the flows into different queues and treating individually. Schedulers of this type can not be scaled as overhead increases as the number of on-going traffic increases. Solution is class-based schedulers such as Constraint Based WFQ and static Priority which schedule traffic in a class-basis fashion. But for the individual flow it would be difficult to get the predictable delay and bandwidth sharing. So care must be taken to apply this to voice application which has strict delay requirements. 3.3.1.3 Loss Recovery We can classify loss recovery into two ways one is Active recovery and the other is Passive recovery. We have retransmission in Active recovery and Forward Error Correction (Adding redundancy) in passive recovery. Retransmission may not be suitable for VoIP because of it latency of packets increases. 3.3.2 Control Plane 3.3.2.1 Resource provisioning and Traffic Engineering Refers to the configuration of resources for applications in the network. In industry, main approach of resource provisioning is over provisioning, abundantly providing resources. Factors that make this attractive are cost of bandwidth and network planning, cost of bandwidth in the backbone is decreasing day by day and network planning is becoming simpler. 3.3.2.2Traffic Engineering It mainly focuses to keep the control on network means to minimize the over-utilization of a particular portion of the network while the capacity is available elsewhere in the network. The two methods used to provide powerful tools for traffic engineering are Multi-Protocol Label Switching (MPLS) and Constraint Based Routing (CBR). These are the mechanisms through which a certain amount of network resources can be reserved for the potential voice traffic along the paths which are determined by Constraint Based Routing or other shortest path routing algorithms. 3.3.2.3 Admission Control Admission control is used to limit the resource usage of voice traffic within the amount of the specified resources. There is no provision of admission control in IP networks so it can offer only best effort service. Parameter based Admission Control provides delay guaranteed service to applications which can be accurately described, such as VoIP. In case of bursty traffic, it is difficult to describe traffic characteristics which makes this type to overbook network resources and therefore lowers network utilization. To limit the amount of traffic over any period it uses explicit traffic descriptors (typical example is token bucket). Different algorithms used in parameter based admission control are: ÃâÃÅ" Ciscos resource reservation based (RSVP). ÃâÃÅ" Utilization based (compares with a threshold, based on utilization value at runtime it decides to admit or reject). ÃâÃÅ" Per-flow end-to-end guaranteed delay service (Computes bandwidth requirements and compares with available resource to make decision). ÃâÃÅ" Class-based admission control. 3.4 Performance Evaluation in VoIP applications 3.4.1 End-To-End Delay When End to End delay exceeds a certain value, the interactive ness becomes more like a half-duplex communication. There can be of two type of delay: 1) Delays due to processing and transmission of speech 2) Network delay (delay that is the result of processing with in the system) Network delay = Fixed part + variable part Fixed part depends upon the performance of the network nodes on the transmission path, transmission and propagation delay and the capacity of links between the nodes. Variable part is the time spent in the queues which depends on the network load. Queuing delay can be minimized by using the advanced scheduling mechanisms e.g. Priority queuing. IP packet delay can be reduced by sending shorter packets instead of longer packets. Useful technique for voice delay reduction on WAN is link fragmentation and interleaving. Fragment the lower packet into smaller packets and between those small packets VOICE packets are sent. 3.4.2 Delay Jitter Delay variation, also known as jitter, creates hurdle in the proper reconstruction of voice packets in their original sequential form. It is defined as difference in total end-to-end delay of two consecutive packets in the flow. In order to remove jitter, it requires collecting and storing packets long enough to permit the slowest packets to arrive in order to be played in the correct sequence. Solution is to employ a play out buffer at the receiver to absorb the jitter before outputting the audio stream. Packets are buffered until their scheduled play out time arrives. Scheduling a later deadline increases the possibility of playing out more packets and results in lower loss rate, but at the cost of higher buffering delay. Techniques for Jitter Absorption â⬠¢ Setting the same play out time for all the packets for entire session or for the duration of each session. â⬠¢ Adaptive adjusting of play out time during silence periods regarding to current network â⬠¢ Constantly adapting the play out time for each packet, this requires the scaling of voice packets to maintain continued play out. 3.4.3 Frame Eraser (F.E) It actually happens at that time when the IP packet carrying speech frame does not arrive at the receiver side in time. There may be loss of single frame or a block of frames. Techniques used to encounter the frame erasure â⬠¢ Forward Error Correction (requires additional processing) depends on the rate and distribution of the losses. â⬠¢ Loss concealment (replaces lost frames by playing the last successfully received frame) effective only at low loss rate of a single frame. High F.E and delays can become troublesome because it can lead to a longer period of corrupt voice. The speech quality perceived by the listener is based on F.E levels that occur on the exit from the jitter buffer after the Forward Error Correction has been employed. To reduce levels of frame loss, Assured forwarding service helps to reduce network packet loss that occur because of full queues in network nodes. 3.4.4 Out of Order Packet Delivery This type of problem occurs in the complex topology where number of paths exists between the sender and the receiver. At the receiving end the receiving system must rearrange received packets in the correct order to reconstruct the original speech signal. Techniques for OUT-OF-ORDER PACKET DELIVERY It is also done by Jitter buffer whose functionality now became â⬠¢ Re-ordering out of order packets ( based on sequence number) â⬠¢ Elimination of Jitter Analysis of QoS Parameters Analysis of QoS Parameters Chapter 3 3. Analysis of QoS Parameters 3.1 Introduction A Number of QoS [11] of parameters can be measured and monitored to determine whether a service level offered or received is being achieved. These parameters consist of the following 1. Network availability 2. Bandwidth 3. Delay 4. Jitter 5. Loss 3.1.1 Network Availability Network availability can have a consequential effect on QoS. Simply put, if the network is not available, even during short periods of time, the user or application may achieve unpredictable or undesirable performance (QoS) [11]. Network availability is the summation of the availability of many items that are used to create a network. These include network device redundancy, e.g. redundant interfaces, processor cards or power supplies in routers and switches, resilient networking protocols, multiple physical connections, e.g. fiber or copper, backup power sources etc. Network operators can increase their networks availability by implementing varying degrees of each item. 3.1.2 Bandwidth Bandwidth is one of the most important QoS parameter. It can be divided in to two types 1. Guaranteed bandwidth 2. Available bandwidth 3.1.2.1 Guaranteed bandwidth Network operators offer a service that provides minimum BW and burst BW in the SLA. Because the guaranteed BW the service costs higher as compare to the available BW service. So the service providers must ensure the special treatment to the subscribers who have got the guaranteed BW service. The network operator separates the subscribers by different physical or logical networks in some cases, e.g., VLANs, Virtual Circuits, etc. In some cases, the guaranteed BW service traffic may share the same network infrastructure with available BW service traffic. We often use to see the case at location where network connections are expensive or the bandwidth is leased from another service provider. When subscribers share the same network infrastructure, the subscribers of the guaranteed BW service must get the priority over the available BW subscribers traffic so that in times of networks congestion the guaranteed BW subscribers SLAs are met. Burst BW can be specified in terms of amount and du ration of excess BW (burst) above the guaranteed minimum. QoS mechanism may be activated to avoid or discard traffic that use consistently above the guaranteed minimum BW that the subscriber agreed to in the SLA. 3.1.2.2 Available bandwidth As we know network operators have fixed Bandwidth, but to get more return on the investment of their network infrastructure, they oversubscribe the BW. By oversubscribing the BW a user is subscribed to be no always available to them. This allows users to compete for available BW. They get more or less BW it depends upon the amount of traffic form other users on the network at any given time. Available bandwidth is a technique commonly used over consumer ADSL networks, e.g., a customer signs up for a 384-kbps service that provides no QoS (BW) guarantee in the SLA. The SLA points out that the 384-kbps is standard but does not make any guarantees. Under lightly loaded conditions, the 384-kbps BW will be available to the users but upon network loaded condition, this BW will not be available consistently. It can be noticed during certain times of the day when number of users access the network. 3.1.3 Delay Network delay is the transit time an application experiences from the ingress (entering) point to the egress (exit) point of the network. Delay can cause significant QoS issues with application such as Video conferencing and fax transmission that simply time-out and final under excessive delay conditions. Some applications can compensate for small amounts of delay but once a certain amount is exceeded, the QoS becomes compromised. For example some networking equipment can spoof an SNA session on a host by providing local acknowledgements when the network delay would cause the SNA session to time out. Similarly, VoIP gateways and phones provide some local buffering to compensate for network delay. There can be both fixed and variable delays. Examples of fixed delays are: Application based delay, e.g., voice codec processing time and IP packet creation time by the TCP/IP software stack Data transmission (queuing delay) over the physical network media at each network hop. Propagation delay across the network based on transmission distance Examples of variable delays are: â⬠¢ Ingress queuing delay for traffic entering a network node â⬠¢ Contention with other traffic at each network node â⬠¢ Egress queuing delay for traffic exiting a network node 3.1.4 Jitter (Delay Variation) Jitter is the difference in delay presented by different packets that are part of the same traffic flow. High frequency delay variation is known as jitter and the low frequency delay variation is known as wander. Primary cause of jitter is basically the differences in queue wait times for consecutive packets in a flow and this is the most significant issue for QoS. Traffic types especially real time traffic such as video conferencing can not tolerate jitter. Differences in packet arrival times cause in the voice. All transport system exhibit some jitter. As long as jitter limits below the defined tolerance level, it does not affect service quality. 3.1.5 Loss Loss either bit errors or packet drops has a significant impact on VoIP services as compare to the data services. During the transmission of the voice, loss of multiple packets may cause an audible pop that will become irritating to the user. Now as compare to the voice transmission, in data transmission loss of single bit or multiple packets of information will not effect the whole communication and is almost never noticed by users. In case of real time video conferencing, consecutive packet loss may cause a momentary glitch (defect) on the screen, but the video then proceeds as before. However, if packet drops get increase, then the quality of the transmission degrades. For minimum quality rate of packet loss must be less than 5% and less then 1% for toll quality. When the network node will be congested, it will drop the packets and by this the loss will occur. TCP (Transmission Control Protocol) is one of the networking protocols that offer packets loss protection by the retransmission of packets that may have been dropped by the network. When network congestion will be increased, more packets will be dropped and hence there will be more TCP transmission. If congestion continues the network performance will obviously degrade because much of the BW is being used for the retransmission of dropped packets. TCP will eventually reduce its transmission window size, due to this reduction in window size smaller packets will be transmitted; this will eventually reduce congestion, resulting in fewer packets being dropped. Because congestion has a direct influence on packet loss, congestion avoidance mechanism is often deployed. One such mechanism is called Random Early Discard (RED). RED algorithms randomly and intentionally drop packets once the traff ic reaches one or more configured threshold. RED provides more efficient congestion management for TCP-based flows. 3.1.5.1 Emission priorities It determines the order in which traffic is transmitted as it exits a network node. Traffic with higher emission priority is transmitted a head of traffic with a lower emission priority. Emission priorities also determine the amount of latency introduced to the traffic by the network nodes queuing mechanism. For example, email which is a delay tolerant application will get the lower emission priority as compare to the delay sensitive real time applications such as voice or video. These delay sensitive applications can not be buffered but are being transmitted while the delay tolerant applications may be buffered. In a simple way we can say that emission priorities use a simple transmit priority scheme whereby higher emission priority traffic is always transmitted ahead of lower emission priority traffic. This is typically accomplished using strict priority scheduling (queuing) the downside of this approach is that low emission priority queues may never get services (starved) it there is always higher emission priority traffic with no BW rate limiting. A more detailed scheme provides a weighted scheduling approach to the transmission of the traffic to improve fairness, i.e., the lower emission priority traffic is transmitted. Finally, some emission priority schemes provide a mixture of both priority and weighted schedulers. 3.1.5.2 Discarded priorities Are used to determine the order in which traffic gets discarded. Due to the network congestion packets may be get dropped i.e., the traffic exceeds its prescribed amount of BW for some period of time. When the network will be congested, traffic with a higher discard priority will get drop as compare to the traffic with a lower discard priority. Traffic with similar QoS performance can be sub divided using discard priorities. This allows the traffic to receive the same performance when the network node is not congested. However, when the network node gets congested, the discard priority is used to drop the more suitable traffic first. Discard priorities also allow traffic with the same emission priority to be discarded when the traffic is out of profile. With out discard priorities traffic would need to be separated into different queues in a network node to provide service differentiation. This can be expensive since only a limited number of hardware queues (typically eight or less) are available on networking devices. Some devices may have software based queues but as these are increasingly used, network node performance is typically reduced. With discard priorities, traffic can be placed in the same queue but in effect the queue is sub divided into virtual queues, each with a different discard priority. For example if a product supports three discard priorities, then one hardware queues in effect provides three QoS Levels. Performance Dimension Application Bandwidth Sensitivity to Delay Jitter Loss VoIP Low High High Medium Video Conf High High High Medium Streaming Video on Demand High Medium Medium Medium Streaming Audio Low Medium Medium Medium Client Server Transaction Medium Medium Low High Email Low Low Low High File Transfer Medium Low Low High Table 3.1: Application performance dimensions (use histogram) Table 3.1 illustrates the QoS performance dimensions required by some common applications. Applications can have very different QoS requirements. As these are mixed over a common IP transport network, without applying QoS the network traffic will experience unpredictable behavior. 3.2 Categorizing Applications Networked applications can be categorized based on end user application requirements. Some applications are between people while other applications are a person and a networked device application, e.g., a PC and web server. Finally, some networking devices, e.g., router-to-router. Table 3.2 categorizes applications into four different traffic categories: 1. Network Control 2. Responsive 3. Interactive 4. Timely Traffic Category Example Application Network Control Critical Alarm, routing, billing ETC. Responsive Streaming Audio/Video, Client/Server Transaction Interactive VoIP, Interactive gaming, Video Conferencing Timely Email, Non Critical Table 3.2: Application Categorization 3.2.1 Network Control Applications Some applications are used to control the operations and administration of the network. Such application include network routing protocols, billing applications and QoS monitoring and measuring for SLAs. These applications can be subdivided into those required for critical and standard network operating conditions. To create high availability networks, network control applications require priority over end user applications because if the network is not operating properly, end user application performance will suffer. 3.2.2 Responsive applications Some applications are between a person and networked devices applications to be responsive so a quick response back to the sender (source) is required when the request is being sent to the networking device. Sometimes these applications are referred to as being near real time. These near real time applications require relatively low packet delay, jitter and loss. However QoS requirements for the responsive applications are not as stringent as real time, interactive application requirements. This category includes streaming media and client server web based applications. Streaming media application includes Internet radio and audio / video broadcasts (news, training, education and motion pictures). Streaming applications e.g. videos require the network to be responsive when they are initiated so the user doesnt wait for long time before the media begins playing. For certain types of signaling these applications require the network to be responsive also. For example with movie on deman d when a user changes channels or forward, rewinds or pause the media user expects the application to react similarly to the response time of there remote control. The Client / server web applications typically involve the user selecting a hyperlink to jump from one page to another or submit a request etc. These applications also require the network to be responsive such that once the hyperlink to be responsive such that once the hyperlink is selected, a response. This can be achieved over a best effort network with the help of broadband internet connection as compare to dial up. Financial transaction may be included in these types of application, e.g., place credit card order and quickly provide feedback to the user indicating that either the transaction has completed or not. Otherwise the user may be unsure to initiate a duplicate order. Alternatively the user may assume that the order was placed correctly but it may not have. In either case the user will not be satisfied with the network or applications performance. Responsive applications can use either UDP or TCP based transport. Streaming media applications typically use UDP because in UDP it would not be fruitful to retransmit the data. Web based applications are based on the hypertext transport protocol and always use TCP, for web based application packet loss can be managed by transmission control protocol (TCP) which retransmit lost packets. In case of retransmission of lost streaming media is sufficiently buffered. If not then the lost packets are discarded. This results in the form of distortion in media. 3.2.3 Interactive Applications Some applications are interactive whereby two or more people communicate or participate actively. The participants expect the real time response from the networked applications. In this context real time means that there is minimal delay (latency) and delay variations (jitter) between the sender and receiver. Some interactive applications, such as a telephone call, have operated in real time over the telephone companies circuit switched networks for over 100 years. The QoS expectations for voice applications have been set and therefore must also be achieved for packetized voice such as VoIP. Other interactive applications include video conferencing and interactive gaming. Since the interactive applications operate in real time, packet loss must be minimized. Interactive applications typically are UDP based (Universal Datagram Protocol) and hence cannot retransmit lost or dropped packets as with TCP based applications. However it would not be beneficial to retransmit the packets because interactive applications are time based. For example if a voice packet was lost. It doesnt make sense to retransmit the packet because the conservations between the sender and receiver have already progressed and the lost packet might be from part of the conversation that has already passed in time. 3.2.4 Timely Applications There are some applications which do not require real time performance between a person and networked devices application but do require the information to be delivered in a timely manner. Such example includes save and send or forward email applications and file transfer. The relative importance of these applications is based on their business priorities. These applications require that packets arrive with abounded amount of delay. For example, if an email takes few minutes to arrive at its destination, this is acceptable. However if we consider it in a business environment, if an email takes 10 minutes to arrive at its destination, this will often not acceptable. The same bounded delay applies to file transfer. Once a file transfer is initiated, delay and jitter are illogical because file transfer often take minutes to complete. It is important to note that timely applications use TCP based transport instead of UDP based transport and therefore packet loss is managed by TCP which r etransmit any lost packets resulting in no packet loss. By summarizing above paragraph we can say that timely applications expect the network QoS to provide packets with a bounded amount of delay not more than that. Jitter has a negligible effect on these types of applications. Loss is reduced to zero due to TCPs retransmission mechanism. 3.3 QoS Management Architecture We can divide QoS management architecture of VoIP into two planes: data plane and control plane. Packet classification, shaping, policing, buffer management, scheduling, loss recovery, and error concealment are involved in the mechanism of data plane. They implement the actions the network needs to take on user packets, in order to enforce different class services. Mechanisms which come in control plane are resource provisioning, traffic engineering, admission control, resource reservation and connection management etc. 3.3.1 Data Plane 3.3.1.1 Packet Forwarding It consists of Classifier, Marker, Meter, Shaper / Dropper. When a packet is received, a packet classifier is used to determine which flow or class the packet belongs to. Those packets belong to the same flow/class obey a predefined rule and are processed in an alike manner. The basic criteria of classification for VoIP applications could be IP address, TCP/UDP port, IP precedence, protocol, input port, DiffServ code points (DSCP), or Ethernet 802.1p class of service (CoS). Cisco supports several additional criteria such as access list and traffic profile. The purpose of the meter is to decide whether the packet is in traffic profile or not. The Shaper/Dropper drops the packets which crossed the limits of traffic profile to bring in conformance to current network load. A marker is used to mark the certain field in the packet, such as DS field, to label the packet type for differential treatment later. After the traffic conditioner, buffer is used for packet storage that waits for transmission. 3.3.1.2 Buffer Management and Scheduling Active queue management (RED) drops packets before the repletion of the queue can avoid the problem of unfair resource usage. Predictable queuing delay and bandwidth sharing can be achieved by putting the flows into different queues and treating individually. Schedulers of this type can not be scaled as overhead increases as the number of on-going traffic increases. Solution is class-based schedulers such as Constraint Based WFQ and static Priority which schedule traffic in a class-basis fashion. But for the individual flow it would be difficult to get the predictable delay and bandwidth sharing. So care must be taken to apply this to voice application which has strict delay requirements. 3.3.1.3 Loss Recovery We can classify loss recovery into two ways one is Active recovery and the other is Passive recovery. We have retransmission in Active recovery and Forward Error Correction (Adding redundancy) in passive recovery. Retransmission may not be suitable for VoIP because of it latency of packets increases. 3.3.2 Control Plane 3.3.2.1 Resource provisioning and Traffic Engineering Refers to the configuration of resources for applications in the network. In industry, main approach of resource provisioning is over provisioning, abundantly providing resources. Factors that make this attractive are cost of bandwidth and network planning, cost of bandwidth in the backbone is decreasing day by day and network planning is becoming simpler. 3.3.2.2Traffic Engineering It mainly focuses to keep the control on network means to minimize the over-utilization of a particular portion of the network while the capacity is available elsewhere in the network. The two methods used to provide powerful tools for traffic engineering are Multi-Protocol Label Switching (MPLS) and Constraint Based Routing (CBR). These are the mechanisms through which a certain amount of network resources can be reserved for the potential voice traffic along the paths which are determined by Constraint Based Routing or other shortest path routing algorithms. 3.3.2.3 Admission Control Admission control is used to limit the resource usage of voice traffic within the amount of the specified resources. There is no provision of admission control in IP networks so it can offer only best effort service. Parameter based Admission Control provides delay guaranteed service to applications which can be accurately described, such as VoIP. In case of bursty traffic, it is difficult to describe traffic characteristics which makes this type to overbook network resources and therefore lowers network utilization. To limit the amount of traffic over any period it uses explicit traffic descriptors (typical example is token bucket). Different algorithms used in parameter based admission control are: ÃâÃÅ" Ciscos resource reservation based (RSVP). ÃâÃÅ" Utilization based (compares with a threshold, based on utilization value at runtime it decides to admit or reject). ÃâÃÅ" Per-flow end-to-end guaranteed delay service (Computes bandwidth requirements and compares with available resource to make decision). ÃâÃÅ" Class-based admission control. 3.4 Performance Evaluation in VoIP applications 3.4.1 End-To-End Delay When End to End delay exceeds a certain value, the interactive ness becomes more like a half-duplex communication. There can be of two type of delay: 1) Delays due to processing and transmission of speech 2) Network delay (delay that is the result of processing with in the system) Network delay = Fixed part + variable part Fixed part depends upon the performance of the network nodes on the transmission path, transmission and propagation delay and the capacity of links between the nodes. Variable part is the time spent in the queues which depends on the network load. Queuing delay can be minimized by using the advanced scheduling mechanisms e.g. Priority queuing. IP packet delay can be reduced by sending shorter packets instead of longer packets. Useful technique for voice delay reduction on WAN is link fragmentation and interleaving. Fragment the lower packet into smaller packets and between those small packets VOICE packets are sent. 3.4.2 Delay Jitter Delay variation, also known as jitter, creates hurdle in the proper reconstruction of voice packets in their original sequential form. It is defined as difference in total end-to-end delay of two consecutive packets in the flow. In order to remove jitter, it requires collecting and storing packets long enough to permit the slowest packets to arrive in order to be played in the correct sequence. Solution is to employ a play out buffer at the receiver to absorb the jitter before outputting the audio stream. Packets are buffered until their scheduled play out time arrives. Scheduling a later deadline increases the possibility of playing out more packets and results in lower loss rate, but at the cost of higher buffering delay. Techniques for Jitter Absorption â⬠¢ Setting the same play out time for all the packets for entire session or for the duration of each session. â⬠¢ Adaptive adjusting of play out time during silence periods regarding to current network â⬠¢ Constantly adapting the play out time for each packet, this requires the scaling of voice packets to maintain continued play out. 3.4.3 Frame Eraser (F.E) It actually happens at that time when the IP packet carrying speech frame does not arrive at the receiver side in time. There may be loss of single frame or a block of frames. Techniques used to encounter the frame erasure â⬠¢ Forward Error Correction (requires additional processing) depends on the rate and distribution of the losses. â⬠¢ Loss concealment (replaces lost frames by playing the last successfully received frame) effective only at low loss rate of a single frame. High F.E and delays can become troublesome because it can lead to a longer period of corrupt voice. The speech quality perceived by the listener is based on F.E levels that occur on the exit from the jitter buffer after the Forward Error Correction has been employed. To reduce levels of frame loss, Assured forwarding service helps to reduce network packet loss that occur because of full queues in network nodes. 3.4.4 Out of Order Packet Delivery This type of problem occurs in the complex topology where number of paths exists between the sender and the receiver. At the receiving end the receiving system must rearrange received packets in the correct order to reconstruct the original speech signal. Techniques for OUT-OF-ORDER PACKET DELIVERY It is also done by Jitter buffer whose functionality now became â⬠¢ Re-ordering out of order packets ( based on sequence number) â⬠¢ Elimination of Jitter
socio-economic development and health Essay -- essays research papers
Question One There are a number of ways in which the increasing socio-economic development of a nation can help improve the health of the population. 1.à à à à à There is a correlation between mortality rates in the developing countries, especially amongst children, and the level of education of the parents of the children. For example, in Morocco, a mother who has completed 4-6 years of schooling, their child is 45% less likely to have died by the age of 2, compared with childââ¬â¢s mother who has had no school (Book 3, Page 54). Education improves the overall knowledge of looking after oneself and others, but also enables people to gain higher income levels, and thus, acquire purchasing power to buy the goods (if available), which will help them improve their quality of life. 2.à à à à à Food provisions are a necessity to maintaining a healthy population. There are many facets to food, mainly the distribution and supply of food, and the quality and nutritional ingredients of food. Food needs to be of good, sustainable quality so that it provides people with the basic supply of vitamins and minerals to live, and has to be easily accessible so that everyone in the nation can benefit. Developed countries have pioneered the way of preserving food for longer (i.e. use of plastics), and developing countries have benefited from this, but the developed world has also introduced new fear factors regarding food such as contamination (BSE, Salmonella etc) and additives, and, the long term effects of such advancements is beginning to materialise (Book 3, Page 306-307). Developing nations need to maintain a balance of growth, by producing enough food for the nations own consumption, but also growing food for exportation, which will improve their GNP and thei r overall growth as a nation. 3.à à à à à Reducing the gap between the social classes will provide a better overall health and wealth of a nation. Those living in the lower social classes have a lower life expectancy than those in higher social classes (Book 3, Page 216). There are many tools and precautions that may be used to bridge the gap. Occupations within the social classes tend to be more manual and risk-based occupations such as mining or engineering. In recent times, Acts of Law have been passed by Governments to protect employees, and as such... ... in further research. The developed world cannot be complacent in its attitude towards communicable diseases. As more and more people are able and free to roam from country to country, so it becomes harder to ensure that adequate strategies can be enforced and that the appropriate vaccines have been administered. Therefore, there still has to be concerted efforts from the developed and the developing world that a multi-disciplinary strategy can be adopted and enforced, and only by such mechanisms can the long-term goal of eradicating communicable diseases be achieved. References Szreter,S (1998) ââ¬ËThe importance of social intervention in Britainââ¬â¢s mortality decline c. 1850-1914: a re-interpretation of the role of public health.ââ¬â¢ in Davey, B, Gray, A and Seale, C (eds) Health and Disease: A Reader, Open University Press, Buckingham. U205 Health and Disease Book 3 (2001) World Health and Disease, Gray,A, Open University Press, Buckingham. U205 Health and Disease Book 1 (2001) Medical Knowledge: Doubt and Certainty, Seale, C. Pattison, S. and Davey, B. (eds), Open University Press, Buckingham. VC 1265, Video 1, ââ¬ËSouth Africa: Health at the crossroadsââ¬â¢ Open University.
Tuesday, October 1, 2019
Effects of study habits in relation to the academic performance Essay
Introduction Education plays a vital role into every student, especially in our current situation where those who finish with degree are the only ones who has a chance of getting hired. Before even getting hired, people must first finish their studies. Students must survive through college but it is not as easy as it seems to be. They must accomplish all the tasks given in a limited amount of time that is why a study habit is needed. Study habits are the ways a student study. These are the habits that students develop while studying. They can be good ones or bad ones. Study habits are considered as one of the major factor affecting the studentââ¬â¢s academic performance. It means that if a student possesses an ineffective study habit, he will not have a clear understanding in his subject which will most likely lead him to failure. If a student develops an effective study habits then he has a higher chance of passing. The researcher came up with this study since she herself does not have an effective study habit and always cram whenever the time to study is almost over. The researcher decided to find the effective study habits that most students prefer so that it will not be hard for them to survive through college works. Significance of the Study Through this research, students will become aware of the effects of study habits on their grades. The researcher believes this will be beneficial to the school administrators, teachers, parents, students, and to the future researchers. Moreover, the researcher believes that the student, especially the students in Section IJ will be benefiting from this study since it will provide a better understanding on how their study habits will be effective. To the School Administrators, this will help them to know and to inform the teachers on how they can effectively teach their students. To the Teachers, this will serve as a guide for them to teach more effectively in a way that all students will understand. To the Parents, this will serve as a guide for them on how they are going to help their children when studying or preparing for examinations.à To the Students, this will help them a lot not only in studying but also for their future.à To the future researchers, they may be able to use the result of the study in further research similar to what the study is. General Objective This study aims to determine the significant relationship between the different study habits and the academic performance of Section IJ students. Specific Objectivesà 1. To know the different study habits commonly practiced by the Section IJ students. 2. To identify the effects of the common study habits practiced by the Section IJ students. 3. To determine the significant relationship between Section IJ Studentââ¬â¢s academic performance and study habits. Statement of the Problem This study aims to determine the effects of study habits in relation to the academic performance of Section IJ students. Sub-problemsà 1. What are the different study habits commonly practiced by the Section IJ students? 2. What are the effects of the common study habits practiced by the Section IJ students? 3. Is there a significant relationship between the study habits and the academic performance of Section IJ students? Scope and Limitations The study focuses on students of Section IJ so that they will be prepared more in the incoming tests. IJ students are college students that is why they are given more works than before. Having a study habit that suits them well will help finish their works faster. The researcher limits the study to the common study habits practiced by IJ students only. It does not matter whether the study habits they developed is good or bad as long as it has a good effect in their academic performance. Hypothesis There is a significant relationship between the study habits and academic performance of Section IJ students.
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