Design of English Learning Effectiveness Evaluation System Based on K-Means Clustering Algorithm

English is the universal language of the world. In the context of global economic integration, English learning is not only an essential course for business elites but also a required course for the general public. Currently, in colleges and universities across the world, English is presented as a compulsory first foreign language course. Therefore, how to improve the effect of English performance assessment in the context of smart teaching has become an important part of smart English teaching. Due to the influence of interference factors, human factors, or external factors, the traditional English language teaching evaluation system has the problems of high system sensitivity, long envelope delay jitter time, and short stationary state maintenance time. Therefore, this study develops an English learning effectiveness evaluation system based on a K-means clustering algorithm. The SQL Server 2005 database management software is used to develop the system database; various functional modules of the system are designed using ActiveX, with emphasis on the design of scoring functional modules; and different roles and permissions are given to administrators, teachers, and students. A student English learning effectiveness evaluation model based on BP neural network training and K-means clustering algorithm is designed to optimize the English learning effectiveness evaluation model and achieve effective English learning by solving the consistent estimate of the effectiveness of English learning assessment. The performance test results show that the proposed system has a lower sensitivity coefficient, a shorter envelope delay jitter time, and a longer period of steady-state maintenance, indicating that the system can achieve stable operation.

Performance of VoIP Using Routing Open Shortest Path First with Multi-Protocol Label Switching

Voice over internet protocol is a communication technology in the world of computer network that can be used for sending voice or video and data transmission over Internet Protocol in real time. VoIP network can be implemented with Asterisk applications as a server to a Private Automatic Branch eXchange applied in a Graphical Network Simulator 3. In this study, VoIP communication using routing OSPF within MPLS will be calculated, the QoS value collected based on impact performance under normal condition and not normal condition (link failure) different varian bandwidth in the network. The results from the simulation show that in normal condition and not normal condition there is average delay value with routing OSPF 4 ms and routing OSPF with MPLS 5 ms, the value of jitter max which same of 6 ms using varian bandwidth from 256 kbps and 512 kbps. All of the QoS parameters, such as delay, jitter and packet loss will be compare to standard ITU-T G.114. This research can be extended with addition of another measurement or another protocol.

ANALISIS PERBANDINGAN GATEWAY BALANCING PADA SOFTWARE DEFINED NETWORK DENGAN ALGORITMA SCHEDULING TAMBAHAN

Pada proses pengiriman paket data pada suatu jaringan, tentu dibutuhkan arsitektur yang memiliki keandalan yang tinggi. Hal ini dibutuhkan agar dapat menjamin kecepatan pengiriman dan tidak adanya data yang gagal dikirimkan ke tujuan. Tersedianya banyak jalur tentu akan meningkatkan availability pada jaringan tersebut, namun tidak menjamin akan meningkatkan kecepatan atau kualitas pengiriman data dan keberhasilan data tersebut sampai ke tujuan. Adanya dua atau lebih jalur memungkinkan akan terjadinya penyebaran trafik yang tidak merata dan penumpukan trafik pada suatu gateway. Untuk menghindari hal ini, maka akan dilakukan metode Gateway Balancing. Pada penelitian ini dilakukan penerapan gateway balancing pada jaringan Software Defined Network menggunakan controller OpenDayLight dengan menerapkan dua algoritma scheduling yaitu algoritma Ant Colony Optimizaion dan algoritma Random Early Detection. Dengan menerapkan gateway balancing pada dua algoritma ini, maka akan dilakukan pengujian performansi trafik dengan mengalirkan UDP Flows. Parameter pengujian yang digunakan adalah Throughput, Delay, Jitter, dan Packeloss. Disimpulkan bahwa perfomansi trafik menggunakan gateway balancing dengan algortima Ant Colony Optimization menawarkan kecepatan data lebih baik dibandingkan algoritma Random Early Detection dengan selisih delay sebesar 4,46%. Sedangkan pada algoritma Random Early Detection, menghasilkan performa yang lebih baik pada keutuhan datanya, dengan selisih 3,55% pada throughput dan 5,85% pada packetloss yang dihasilkan.

Application Layer ARQ Algorithm for Real-Time Multi-Source Data Streaming in UAV Networks

Because of the specific characteristics of Unmanned Aerial Vehicle (UAV) networks and real-time applications, the trade-off between delay and reliability imposes problems for streaming video. Buffer management and drop packets policies play a critical role in the final quality of the video received by the end station. In this paper, we present a reactive buffer management algorithm, called Multi-Source Application Layer Automatic Repeat Request (MS-AL-ARQ), for a real-time non-interactive video streaming system installed on a standalone UAV network. This algorithm implements a selective-repeat ARQ model for a multi-source download scenario using a shared buffer for packet reordering, packet recovery, and measurement of Quality of Service (QoS) metrics (packet loss rate, delay and, delay jitter). The proposed algorithm MS-AL-ARQ will be injected on the application layer to alleviate packet loss due to wireless interference and collision while the destination node (base station) receives video data in real-time from different transmitters at the same time. Moreover, it will identify and detect packet loss events for each data flow and send Negative-Acknowledgments (NACKs) if packets were lost. Additionally, the one-way packet delay, jitter, and packet loss ratio will be calculated for each data flow to investigate the performances of the algorithm for different numbers of nodes under different network conditions. We show that the presented algorithm improves the QoS of the video data received under the worst network connection conditions. Furthermore, some congestion issues during deep analyses of the algorithm’s performances have been identified and explained.

A Network Slicing Framework for UAV-Aided Vehicular Networks

In a fifth generation (5G) vehicular network architecture, several point of access (PoA) types, including both road side units (RSUs) and aerial relay nodes (ARNs), can be leveraged to undertake the service of an increasing number of vehicular users. In such an architecture, the application of efficient resource allocation schemes is indispensable. In this direction, this paper describes a network slicing scheme for 5G vehicular networks that aims to optimize the performance of modern network services. The proposed architecture consists of ground RSUs and unmanned aerial vehicles (UAVs) acting as ARNs enabling the communication between ground vehicular nodes and providing additional communication resources. Both RSUs and ARNs implement the LTE vehicle-to-everything (LTE-V2X) technology, while the position of each ARN is optimized by applying a fuzzy multi-attribute decision-making (fuzzy MADM) technique. With regard to the proposed network architecture, each RSU maintains a local virtual resource pool (LVRP) which contains local RBs (LRBs) and shared RBs (SRBs), while an SDN controller maintains a virtual resource pool (VRP), where the SRBs of the RSUs are stored. In addition, each ARN maintains its own resource blocks (RBs). For users connected to the RSUs, if the remaining RBs of the current RSU can satisfy the predefined threshold value, the LRBs of the RSU are allocated to user services. On the contrary, if the remaining RBs of the current RSU cannot satisfy the threshold, extra RBs from the VRP are allocated to user services. Similarly, for users connected to ARNs, the satisfaction grade of each user service is monitored considering both the QoS and the signal-to-noise plus interference (SINR) factors. If the satisfaction grade is higher than the predefined threshold value, the service requirements can be satisfied by the remaining RBs of the ARN. On the contrary, if the estimated satisfaction grade is lower than the predefined threshold value, the ARN borrows extra RBs from the LVRP of the corresponding RSU to achieve the required satisfaction grade. Performance evaluation shows that the suggested method optimizes the resource allocation and improves the performance of the offered services in terms of throughput, packet transfer delay, jitter and packet loss ratio, since the use of ARNs that obtain optimal positions improves the channel conditions observed from each vehicular user.

IRTS: An Intelligent and Reliable Transmission Scheme for Screen Updates Delivery in DaaS

Desktop-as-a-service (DaaS) has been recognized as an elastic and economical solution that enables users to access personal desktops from anywhere at any time. During the interaction process of DaaS, users rely on screen updates to perceive execution results remotely, and thus the reliability and timeliness of screen updates transmission have a great influence on users’ quality of experience (QoE). However, the efficient transmission of screen updates in DaaS is facing severe challenges: most transmission schemes applied in DaaS determine sending strategies in terms of pre-set rules, lacking the intelligence to utilize bandwidth rationally and fit new network scenarios. Meanwhile, they tend to focus on reliability or timeliness and perform unsatisfactorily in ensuring reliability and timeliness simultaneously, leading to lower transmission efficiency of screen updates and users’ QoE when network conditions turn unfavorable. In this article, an intelligent and reliable end-to-end transmission scheme (IRTS) is proposed to cope with the preceding issues. IRTS draws support from reinforcement learning by adopting SARSA, an online learning method based on the temporal difference update rule, to grasp the optimal mapping between network states and sending actions, which extricates IRTS from the reliance on pre-set rules and augments its adaptability to different network conditions. Moreover, IRTS guarantees reliability and timeliness via an adaptive loss recovery method, which intends to recover lost screen updates data automatically with fountain code while controlling the number of redundant packets generated. Extensive performance evaluations are conducted, and numerical results show that IRTS outperforms the reference schemes in display quality, end-to-end delay/delay jitter, and fairness when transferring screen updates under various network conditions, proving that IRTS can enhance the transmission efficiency of screen updates and users’ QoE in DaaS.

Single Board Computer Applications as Multi-Server VoIP

Telecommunication technology is developing along with information technology and several innovations in several audio and data transmission and reception techniques. Innovation and communication technology are hoped to be able to create efficiencies in regards to time, equipment, and cost. The Public Switched Telephone Network (PSTN) telephone technology has experienced integration towards communication using Internet Protocol (IP) networks, better known as Voice over Internet Protocol (VoIP). VoIP Technology transmits conversations digitally through IP-based networks, such as internet networks, Wide Area Networks (WAN), and Local Area Networks (LAN). However, the VoIP cannot fully replace PSTN due to several weaknesses, such as delay, jitter, packet loss, as well as security and echo. Telephones calls using VoIP technology are executed using terminals in the form of computer devices or existing analogue telephones. The benefit of VoIP is that it can be set in all ethernet and IP addresses. Prefixes can be applied for inter-server placements as inter-building telephone networks without the addition of inefficient new cables on single board computers with Elastix installed. Prefix and non-prefix analysis on servers from single board computers can be tested using QoS for bandwidth, jitter, and packet loss codec. The installation of 6 clients, or 3 simultaneous calls resulted in a packet loss value in the prefix Speex codex of 2.34%. The bandwidth in the prefix PCMU codec has an average value of 82.3Kbps, and a non-prefix value of 79.3Kbps, in accordance to the codec standards in the VoIP. The lowest jitter was found in the non-prefix PCMU codec with an average of 51.05ms, with the highest jitter for the prefix Speex codec being 314.65ms.

Evaluasi Kinerja Testbed Routing Protocol berbasis NodeMCU ESP8266 pada Perangkat IoT

AbstrakJaringan Ad hoc pada perangkat Internet of Things (IoT) mempunyai sifat yang yang dinamis dengan node pada jaringan yang berperan sebagai router dan bergerak bebas secara random tanpa bantuan infrasturktur komunikasi sehingga topologi berubah sangat cepat seiring dengan perubahan posisi. Perubahan ini sangat mempengaruhi kualitas layanan pada perangkat IoT itu sendiri. Penelitian ini bertujuan untuk mengevaluasi protocol routing yang sudah ada dengan cara mengimplementasikan routing protocol tersebut di perangkat testbed berbasis NodeMCU ESP8266. Hal ini bertujuan untuk memilih protocol routing yang paling optimal sebelum proses implementasi dilaksanakan. Pengujian ini berlaku untuk routing protocol yang sudah ada maupun yang baru. Kinerja protocol jaringan diukur melalui nilai Quality of Service (QoS) ditempatkan pada scenario berbeda yang terdiri dari throughput, delay, jitter, dan packet delivery ratio sesuai dengan perbedaan beban jaringan, mobilitas, dan ukuran jaringan. Hasil penelitian ini menunjukkan bahwa testbed yang dibangun berhasil mensimulasikan routing protocol yang ada untuk menghasilkan QoS yang baik pada perangkat IoT.Kata kunci: IoT, routing protocol, testbed, QoS.AbstractAd hoc networks on Internet of Things (IoT) devices have dynamic characteristics where the nodes on this network can operate as routers and move freely randomly without using any communication infrastructure so that the topology changes very quickly along with changes in position. This adjustment has a significant impact on the IoT device's service quality. This study aims to evaluate the existing routing protocols by implementing the routing protocol in a testbed based on NodeMCU ESP8266. It aims to choose the most optimal routing protocol before the implementation process is carried out. This test applies to both existing and new routing protocols. Network protocol performance is measured by the Quality of Service (QoS) value which includes throughput, delay, jitter, and packet delivery ratio in different scenarios based on network load, mobility, and different network sizes. The results show that this study was successful in simulating routing protocol in order to provide good QoS on IoT devices.Keywords: IoT, routing protocol, testbed, QoS.