7 research outputs found

    Tret personaliti dan faktor pemilihan kategori pekerjaan dalam kalangan pelajar kolej vokasional

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    Kajian ini mengkaji tret personaliti dan faktor pemilihan kategori pekerjaan dalam kalangan pelajar tahun dua Diploma Vokasional Malaysia di Kolej Vokasional dari negeri Johor. Kajian ini menggunakan Teori Tipologi Kerjaya Holland (1973) bagi mengenalpasti tret personaliti dan faktor utama yang mendorong pemilihan kerjaya para pelajar di kolej vokasional. Seramai 104 orang telah menurut serta di dalam kajian ini yang terdiri daripada pelajar tahun 2 Diploma Vokasional Malaysia yang menggunakan persampelan secara rawak. Alat kajian Self�Directed Search (SDS)-Form Eassy digunakan bagi meninjau tret personaliti responden, manakala item bagi faktor yang mendorong pemilihan kategori pekerjaan pelajar dibangunkan sendiri orang pengkaji. Analisis data deskriptif yang diperolehi menunjukkan bahawa faktor utama dalam pemilihan kerjaya responden adalah faktor minat. Dapatan kajian juga menunjukkan profil personaliti keseluruhan responden adalah S(sosial), I(investigatif) dan R(realistik). Ciri-ciri bagi kategori kerjaya ini adalah minat terhadap aktiviti-aktiviti yang berkaitan dengan memanipulasi orang lain seperti memberitahu, melatih, mendidik, menghindari aktiviti yang eksplisit, berstruktur serta aktiviti yang melibatkan bahan, peralatan dan mesin. Analisis Independent Sample T-test digunakan bagi melihat perbezaan yang signifikan di antara faktor yang mempengaruhi pemilihan kerjaya responden. Dapatan kajian menunjukkan terdapat perbezaan yang signifikan di antara faktor minat yang mempengaruhi pemilihan kerjaya responden. Kesimpulan dari kajian ini adalah pemilihan kategori pekerjaan pelajar-pelajar di kolej vokasional adalah SIR iaitu berupaya bekerja dengan orang lain dengan keupayaan kemahiran berfikir yang boleh diaplikasikan dalam kerja-kerja kemahiran bersesuaian bidang pengajian teknologi yang dipilih. Selain daripada itu minat merupakan faktor utama dalam pemilihan bidang dan pelajar-pelajar ini telah bersedia untuk menceburi bidang kerjaya yang selari dengan bidang pengajian. Kepentingan kajian ini boleh menjadi sebahagian dari rujukan bagi pemilihan bidang pengajian kepada pelajar-pelajar yang memilih TVET sebagai laluan kerjaya

    Analisis Kualitas VoIP yang Berjalan di Atas Protokol Datagram Congestion Control Protocol

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    Abstrak Datagram Congestion Control Protocol (DCCP) adalah transport protocol yang didesain untuk aplikasi Internet multimedia. Protocol ini memiliki beberapa mekanisme congestion control dan dapat dipilih sesuai dengan karakteristik aplikasinya. CCID4 adalah salah satu mekanisme congestion control dari DCCP yang sesuai dengan aplikasi VoIP karena digunakan untuk mengirim paket yang berukuran kecil. Selain itu CCID4 mengatur jarak antar paket minimum sepanjang 10 ms sehingga paket VoIP tidak mendominasi penggunaan jaringan. Penelitian ini melakukan analisis terhadap protokol DCCP/CCID4 untuk mengetahui kualitas VoIP yang menggunakan protokol tersebut, membandingkan kualitas yang dihasilkan dengan penggunaan protokol UDP, serta melihat kualitas aliran TCP sebagai akibat dari terdapatnya aliran VoIP pada jaringan. Hasil simulasi memperlihatkan kualitas VoIP cenderung lebih baik dengan menggunakan protokol UDP dibandingkan dengan DCCP/CCID4 ketika tidak terdapat aliran TCP pada jaringan. Sebaliknya, ketika terdapat aliran TCP maka kualitas VoIP dengan protokol DCCP/CCID4 lebih baik dibandingkan dengan UDP. DCCP/CCID4 berhasil mencegah terjadinya congestion collapse dibandingkan dengan penggunaan UDP. Selain itu, kualitas TCP lebih baik ketika protokol DCCP/CCID4 digunakan untuk aplikasi VoIP pada jaringan yang mengalami congestion.   Kata kunci—mekanisme congestion control, Voice over Internet Protocol (VoIP), Datagram Congestion Control Protocol (DCCP), TCP, UDP, NS-2     Abstract Datagram Congestion Control Protocol (DCCP) is a transport protocol designed for multimedia Internet applications. This protocol has several congestion control mechanisms and can be selected according to application characteristics. CCID4 is a congestion control mechanism suitable for use with VoIP applications to send a small packet. CCID4 enforces a minimum interval of 10 milliseconds between data packets. This study conducted an analysis of DCCP/CCID4 protocol to determine the quality of VoIP that uses the protocol, comparing the quality produced by the used of UDP protocol, and see the quality of TCP flow as a result of the presence of the VoIP flow on the network. The simulation results show the quality of VoIP tend to do better by using the UDP protocol compared with DCCP/CCID4 when there is no TCP flow on the network. Conversely, when there are TCP flows, the quality of VoIP with DCCP/CCID4 protocol better than UDP. DCCP/CCID4 managed to prevent the congestion collapse compared with the use of UDP. In addition, the TCP quality is better when DCCP/CCID4 protocol used for VoIP applications that experiencing network congestion.   Keywords— congestion control mechanism, Voice over Internet Protocol (VoIP), Datagram Congestion Control Protocol (DCCP), TCP, UDP, NS-

    Model checking techniques for runtime testing and QoS analysis

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    Los sistemas software y hardware se encuentran cada vez más presentes en nuestras vidas, en multitud de campos de aplicación y de cualquier tamaño. El análisis de estos sistemas es una tarea dura pero necesaria para garantizar que cumplan con sus requisitos. Estos requisitos pueden ser de varios tipos, como evitar comportamientos erróneos u ofrecer un rendimiento satisfactorio. Existen muchas técnicas y herramientas diseñadas para atacar este problema. Por lo general, se aplican distintas técnicas dependiendo del tipo de sistema, fase de desarrollo o tipo de análisis. El model checking es una de estas técnicas de análisis. Un model checker analiza el espacio de estados de un sistema para comprobar si el sistema cumple una propiedad dada. Sin embargo, según aumenta la complejidad del sistema a analizar, su espacio de estados crece rápidamente, hasta llegar a un punto en el que no es factible analizarlo. En esta tesis proponemos una solución integrada basada en model checking para analizar sistemas cuyo comportamiento pueda ser observado en forma de trazas de ejecución. Hemos llamado a esta solución OptySim. Nuestra solución permite acceder a sistemas externos de una forma uniforme, permitiendo realizar distintos tipos de análisis sobre diferentes tipos de sistemas de una forma más homogénea. OptySim trata con un conjunto de trazas de ejecución, que representan un subconjunto del espacio de estados completo del sistema. Para obtener dichas trazas el sistema se ejecuta repetidas veces, posiblemente variando parámetros del sistema de acuerdo a las instrucciones del usuario, generándose una traza por cada ejecución. El contenido de las trazas depende de cada sistema, y además puede variar dependiendo de las necesidades del análisis. Para ello se pueden aplicar una de las proyecciones que se han definido, y que transforman trazas completas en trazas abstractas con una menor, pero suficiente para los propósitos del análisis, cantidad de información. El análisis está guiado por uno o más objetivos establecidos por el usuario, tales como asertos o fórmulas de lógica temporal (LTL), y que le dan al análisis el significado pretendido por el usuario. Los objetivos pueden indicar tanto propiedades deseables del sistema, por ejemplo una meta de rendimiento, como propiedades que no deben ocurrir, por ejemplo una condición de error. OptySim se ha aplicado a varios casos de estudio en varias áreas y con distintos propósitos, para demostrar su utilidad. En primer lugar se ha integrado con el simulador de redes ns-2, para análisis de fiabilidad y rendimiento, optimización de parámetros, y validación y ajuste de modelos. Para el segundo grupo de casos de estudio, se ha integrado con una máquina virtual de Java para analizar programas escritos en dicho lenguaje de programación. En esta ocasión, todos los casos de estudio están enfocados a la depuración de programas

    Performance Assessment of Aggregation and Deaggregation Algorithms in Vehicular Delay-Tolerant Networks

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    Vehicular Delay-Tolerant Networks (VDTNs) are a new approach for vehicular communications where vehicles cooperate with each other, acting as the communication infrastructure, to provide low-cost asynchronous opportunistic communications. These communication technologies assume variable delays and bandwidth constraints characterized by a non-transmission control protocol/ internet protocol architecture but interacting with it at the edge of the network. VDTNs are based on the principle of asynchronous communications, bundleoriented communication from the DTN architecture, employing a store-carryand- forward routing paradigm. In this sense, VDTNs should use the tight network resources optimizing each opportunistic contact among nodes. At the ingress edge nodes, incoming IP Packets (datagrams) are assembled into large data packets, called bundles. The bundle aggregation process plays an important role on the performance of VDTN applications. Then, this paper presents three aggregation algorithms based on time, bundle size, and a hybrid solution with combination of both. Furthermore, the following four aggregation schemes with quality of service (QoS) support are proposed: 1) single-class bundle with N = M, 2) composite-class bundle with N = M, 3) single-class bundle with N > M, and 4) composite-class bundle with N > M, where N is the number of classes of incoming packets and M is the number of priorities supported by the VDTN core network. The proposed mechanisms were evaluated through a laboratory testbed, called VDTN@Lab. The adaptive hybrid approach and the composite-class schemes present the best performance for different types of traffic load and best priorities distribution, respectively

    Resource Allocation in LTE Advanced for QoS and Energy Efficiency

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    Long Term Evolution (LTE) and LTE-Advanced (LTE-A) are establishing themselves as the new standard of 4G cellular networks in Europe and in several other parts of the world. Their success will largely depend on their ability to support Quality of Service for different types of users, at reasonable costs. The quality of service will depend on how effectively the cell bandwidth is shared among the users. The cost will depend – among many other factors – on how effectively we exploit the cell capacity. Being able to exploit bandwidth efficiently postpones the time when network upgrades are required. On the other hand, operation costs also depend on the energy efficiency of the cellular network, which should avoid wasting power when few users are connected. As for bandwidth efficiency, the recent LTE/LTE-A standards introduced MIMO (Multiple Input Multiple Output) transmission modes, which allow both reliability and efficiency to be increased. MIMO can increase the throughput significantly. In a MIMO system, the selection of the MIMO transmission modes (whether Transmission Diversity, Spatial Multiplexing, or Multi-User MIMO) plays a key feature in determining the achievable rate and the error probability experienced by the users. MIMO-unaware scheduling policies, which neglect the transmission mode selection problem, do not perform well under MIMO. In the current literature, few MIMO-aware LTE-A scheduling policies have been designed. However, despite being proposed for LTE-A, these solutions do not take into account some constraints inherent to LTE-A, hence leading to unfeasible allocations. In this work, we propose a new framework for Transmission Mode Selection and Frequency. Domain Packet Scheduling, which is compliant with the constraints of the LTE-A standard. The resource allocation framework accommodates real-time requirements and fairness on demand, while the bulk of the resources are allocated in an opportunistic fashion, i.e. so as to maximize the cell throughput. Our results show that our proposal provides real-time connections with the desired level of QoS, without utterly sacrificing the cell throughput. As far as energy efficiency is concerned, we studied the problem of minimizing the RF power used by the eNodeB, while maintaining the same level of service for the users. We devised a provisioning framework that exploits the Multicast/Broadcast over a Single Frequency Network (MBSFN) mechanism to deactivate the eNodeB on some Transmission Time Intervals (TTI), and computes the minimum-power activation required for guaranteeing a given level of service. Our results show that the provisioning framework is stable, and that it allows significant savings with respect to an always-on policy, with marginal impact on the latency experienced by the users

    A cross-layer middleware architecture for time and safety critical applications in MANETs

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    Mobile Ad hoc Networks (MANETs) can be deployed instantaneously and adaptively, making them highly suitable to military, medical and disaster-response scenarios. Using real-time applications for provision of instantaneous and dependable communications, media streaming, and device control in these scenarios is a growing research field. Realising timing requirements in packet delivery is essential to safety-critical real-time applications that are both delay- and loss-sensitive. Safety of these applications is compromised by packet loss, both on the network and by the applications themselves that will drop packets exceeding delay bounds. However, the provision of this required Quality of Service (QoS) must overcome issues relating to the lack of reliable existing infrastructure, conservation of safety-certified functionality. It must also overcome issues relating to the layer-2 dynamics with causal factors including hidden transmitters and fading channels. This thesis proposes that bounded maximum delay and safety-critical application support can be achieved by using cross-layer middleware. Such an approach benefits from the use of established protocols without requiring modifications to safety-certified ones. This research proposes ROAM: a novel, adaptive and scalable cross-layer Real-time Optimising Ad hoc Middleware framework for the provision and maintenance of performance guarantees in self-configuring MANETs. The ROAM framework is designed to be scalable to new optimisers and MANET protocols and requires no modifications of protocol functionality. Four original contributions are proposed: (1) ROAM, a middleware entity abstracts information from the protocol stack using application programming interfaces (APIs) and that implements optimisers to monitor and autonomously tune conditions at protocol layers in response to dynamic network conditions. The cross-layer approach is MANET protocol generic, using minimal imposition on the protocol stack, without protocol modification requirements. (2) A horizontal handoff optimiser that responds to time-varying link quality to ensure optimal and most robust channel usage. (3) A distributed contention reduction optimiser that reduces channel contention and related delay, in response to detection of the presence of a hidden transmitter. (4) A feasibility evaluation of the ROAM architecture to bound maximum delay and jitter in a comprehensive range of ns2-MIRACLE simulation scenarios that demonstrate independence from the key causes of network dynamics: application setting and MANET configuration; including mobility or topology. Experimental results show that ROAM can constrain end-to-end delay, jitter and packet loss, to support real-time applications with critical timing requirements

    Ns2Voip++, an enhanced module for VoIP simulations

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    In the last ten years, many circuit-switched networks for voice have been replaced with packet-switched ones. Hence, simulating Voice over IP has become of paramount importance in assessing the performance of a network. However, a sound performance analysis should be carried out in conditions which are as close as possible to a real deployment. In this paper we present enhancements to ns2voip [1], a module for simulating realistic VoIP traffic with the ns2 simulator. In detail, we add new features, i.e. a correlated model for packet generation in a two-way conversation and implementing a set of realistic playout buffers to simulate the behavior of the receiver. Our code is available at http://cng1.iet.unipi.it/wiki/index.php/Ns2voip
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