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1
Prabadevi, B., and N. Jeyanthi. "Security Solution for ARP Cache Poisoning Attacks in Large Data Centre Networks." Cybernetics and Information Technologies 17, no. 4 (November 27, 2017): 69–86. http://dx.doi.org/10.1515/cait-2017-0042.
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AbstractThe bridge protocol (Address Resolution Protocol) ARP, integrating Ethernet (Layer 2) and IP protocol (Layer 3) plays a vital role in TCP/IP communication since ARP packet is the first packet generated during any TCP/IP communications and they are the first traffic from the host. In the large data center, as the size of the broadcast domain (i.e., number of hosts on the network) increases consequently the broadcast traffic from the communication protocols like ARP also increases. This paper addresses the problem faced by Layer 2 protocols like insecured communication, scalability issues and VM migration issues. The proposed system addresses these issues by introducing two new types of messaging with traditional ARP and also combat the ARP Cache poisoning attacks like host impersonation, MITM, Distributed DoS by making ARP stateful. The components of the proposed methodology first start the process by decoding the packets, updates the invalid entry made by the user with Timestamp feature and messages being introduced. The system has been implemented and compared with various existing solutions.
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Park, Byungjoo, Ankyu Hwang, and Haniph Latchman. "Design of Optimized Multimedia Data Streaming Management Using OMDSM over Mobile Networks." Mobile Information Systems 2017 (2017): 1–13. http://dx.doi.org/10.1155/2017/2867127.
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Mobility management is an essential challenge for supporting reliable multimedia data streaming over wireless and mobile networks in the Internet of Things (IoT) for location-based mobile marketing applications. The communications among mobile nodes for IoT need to have a seamless handover for delivering high quality multimedia services. The Internet Engineering Task Force (IETF) mobility management schemes are the proposals for handling the routing of IPv6 packets to mobile nodes that have moved away from their home network. However, the standard mobility management scheme cannot prevent packet losses due to longer handover latency. In this article, a new enhanced data streaming route optimization scheme is introduced that uses an optimized Transmission Control Protocol (TCP) realignment algorithm in order to prevent the packet disordering problem whenever the nodes in the IoT environment are communicating with each other. With the proposed scheme, data packets sequence realignment can be prevented, the packet traffic speed can be controlled, and the TCP performance can be improved. The experimental results show that managing the packet order in proposed new scheme remarkably increases the overall TCP performance over mobile networks within the IoT environment thus ensuring the high quality of service (QoS) for multimedia data streaming in location-based mobile marketing applications.
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Zheng, Yani, Gaurav Dhiman, Ashutosh Sharma, Amit Sharma, and Mohd Asif Shah. "An IoT-Based Water Level Detection System Enabling Fuzzy Logic Control and Optical Fiber Sensor." Security and Communication Networks 2021 (August 27, 2021): 1–11. http://dx.doi.org/10.1155/2021/4229013.
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The usage of wireless sensors has become widespread for the collection of data for various Internet of Things (IoT) products. Specific wireless sensors use optical fiber technology as transmission media and lightwave signals as carriers, showing the advantages of antielectromagnetic interference, high sensitivity, and strong reliability. Hence, their application in IoT systems becomes a research hotspot. In this article, multiple optical fiber sensors are constructed as an IoT detection system, and a Transmission Control Protocol (TCP)/Internet Protocol (IP) communication stack is used for the sensor module. Furthermore, design of gateway module, data server, and monitoring module is established in order to run the data server in the Windows system and communicate across the network segments. Furthermore, the optical fiber sensor is connected to the development board with WiFi, meanwhile considering the optical fiber wireless network’s congestion problem. The fuzzy logic concept is introduced from the perspective of cache occupancy, and a fiber sensor’s network congestion control algorithm is proposed. In the experiment, the IoT detection system with multiple optical fiber sensors is used for water level detection, and the sensor’s real-time data detected by the User Interface (UI) are consistent with the feedback results. The proposed method is also compared with the SenTCP algorithm and the CODA algorithm, and it was observed that the proposed network congestion control algorithm based on the fuzzy logic can improve network throughput and reduce the network data packet loss.
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Irawan, Addie, Mohammad Fadhil Abas, and Nurulfadzilah Hasan. "Robot Local Network Using TQS Protocol for Land-to-Underwater Communications." Journal of Telecommunications and Information Technology 1 (March 29, 2019): 23–30. http://dx.doi.org/10.26636/jtit.2019.125818.
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This paper presents a model and an analysis of the Tag QoS switching (TQS) protocol proposed for heterogeneous robots operating in different environments. Collaborative control is topic that is widely discussed in multirobot task allocation (MRTA) – an area which includes establishing network communication between each of the connected robots. Therefore, this research focuses on classifying, prioritizing and analyzing performance of the robot local network (RLN) model which comprises a point-to-point topology network between robot peers (nodes) in the air, on land, and under water. The proposed TQS protocol was inspired by multiprotocol label switching (MPLS), achieving a quality of service (QoS) where swapping and labeling operations involving the data packet header were applied. The OMNET++ discrete event simulator was used to analyze the percentage of losses, average access delay, and throughput of the transmitted data in different classes of service (CoS), in a line of transmission between underwater and land environments. The results show that inferior data transmission performance has the lowest priority with low bitrates and extremely high data packet loss rates when the network traffic was busy. On the other hand, simulation results for the highest CoS data forwarding show that its performance was not affected by different data transmission rates characterizing different mediums and environments.
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Atmadja, Martono Dwi. "Single Board Computer Applications as Multi-Server VoIP." International Journal for Research in Applied Science and Engineering Technology 9, no. VII (July 15, 2021): 1023–28. http://dx.doi.org/10.22214/ijraset.2021.36512.
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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.
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Memon, Sheeba, Jiawei Huang, and Hussain Saajid. "Gentle Slow Start to Alleviate TCP Incast in Data Center Networks." Symmetry 11, no. 2 (January 27, 2019): 138. http://dx.doi.org/10.3390/sym11020138.
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Modern data center networks typically adopt symmetric topologies, such as leaf-spine and fat-tree. When a large number of transmission control protocol (TCP) flows in data center networks send data to the same receiver, the congestion collapse, called TCP Incast, frequently happens because of the huge packet losses and Time-Out. To address the TCP Incast issue, we firstly demonstrate that adjusting the increasing speed of the congestion window during the slow start phase is crucially important. Then we propose the Gentle Slow Start (GSS) algorithm, which adjusts the congestion window according to real-time congestion state in a gentle manner and smoothly switches from slow start to congestion avoidance phase. Furthermore, we present the implementation and design of Gentle Slow Start and also integrate it into the state-of-the-art data center transport protocols. The test results show that GSS effectively decreases the Incast probability and increases the network goodput by average 8x.
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Lee, Sungwon, Yeongjoon Bae, Muhammad Toaha Raza Khan, Junho Seo, and Dongkyun Kim. "Avoiding Spurious Retransmission over Flooding-Based Routing Protocol for Underwater Sensor Networks." Wireless Communications and Mobile Computing 2020 (August 25, 2020): 1–9. http://dx.doi.org/10.1155/2020/8839541.
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In underwater wireless sensor networks (UWSN), acoustic communication naturally introduces challenges such as long propagation delay and high packet loss. The flooding-based routing protocol can address these challenges with its multipath characteristics. As in flooding-based routing, due to multipath propagation mechanism, not only DATA but also ACK messages are transmitted through multiple routes however still some packet loss will degrade the performance. So, to provide high reliability of message delivery, an efficient retransmission mechanism is inevitable. Though, if the network uses conventional transport layer protocol such as TCP, it will suffer a spurious retransmission problem as TCP was originally not designed for the multipath environment. In this paper, we propose route discrimination for flooding-based routing to reduce spurious retransmission in UWSN to solve the limitation. The notion of ACK copies waiting time (ACWT) is utilized which is selectively updated based on the similarity of paths of transmission of ACK message copies. We also improved our previous solution that lacks flexibility to cope with dynamic link error characteristics. Through evaluation, we verified that our new scheme achieves the performance improvements of 14%~84% in terms of retransmission ratio compared to the previous research.
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Xu, Yong, Hong Ni, and Xiaoyong Zhu. "An Effective Transmission Scheme Based on Early Congestion Detection for Information-Centric Network." Electronics 10, no. 18 (September 9, 2021): 2205. http://dx.doi.org/10.3390/electronics10182205.
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As one of the candidates for future network architecture, Information-Centric Networking (ICN) has revolutionized the manner of content retrieval by transforming the communication mode from host-centric to information-centric. Unlike a traditional TCP/IP network, ICN uses a location-independent name to identify content and takes a receiver-driven model to retrieve the content. Moreover, ICN routers not only perform a forwarding function but also act as content providers due to pervasive in-network caching. The network traffic is more complicated and routers are more prone to congestion. These distinguished characteristics pose new challenges to ICN transmission control mechanism. In this paper, we propose an effective transmission scheme by combining the receiver-driven transport protocol and the router-driven congestion detection mechanism. We first outline the process of content retrieval and transmission in an IP-compatible ICN architecture and propose a practical receiver-driven transport protocol. Then, we present an early congestion detection mechanism applied on ICN routers based on an improved Active Queue Management (AQM) algorithm and design a receiver-driven congestion control algorithm. Finally, experiment results show that the proposed transmission scheme can maintain high bandwidth utilization and significantly reduce transmission delay and packet loss rate.
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Reddy, N. Ramanjaneya, Chenna Reddy Pakanati, and M. Padmavathamma. "An Enhanced Queue Management Scheme for Eradicating Congestion of TFRC over Wired Environment." International Journal of Electrical and Computer Engineering (IJECE) 7, no. 3 (June 1, 2017): 1347. http://dx.doi.org/10.11591/ijece.v7i3.pp1347-1354.
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<p class="Abstract">To accomplish increasing real time requirements, user applications have to send different kinds of data with different speeds over the internet. To effectuate the aims of the computer networks, several protocols have been added to TCP/IP protocol suite. Transport layer has to implement emerging techniques to transfer huge amount of data like multimedia streaming. To transmit multimedia applications, one of the suitable congestion control mechanisms in transport layer is TCP Friendly Rate Control Protocol (TFRC). It controls congestion based on its equation. To get more smoothed throughput, intermediate nodes (like Routers. etc.) have to use suitable procedures in all real time situations. To eradicate the level of congestion in the network, we introduce enhanced Holt-Winters equations to RED queue management algorithm and applied to TFRC. The simulation results have shown that this strategy reduces packet loss and increases throughput.</p>
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Mudassir, Mumajjed Ul, and M. Iram Baig. "MFVL HCCA: A Modified Fast-Vegas-LIA Hybrid Congestion Control Algorithm for MPTCP Traffic Flows in Multihomed Smart Gas IoT Networks." Electronics 10, no. 6 (March 18, 2021): 711. http://dx.doi.org/10.3390/electronics10060711.
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Multihomed smart gas meters are Internet of Things (IoT) devices that transmit information wirelessly to a cloud or remote database via multiple network paths. The information is utilized by the smart gas grid for accurate load forecasting and several other important tasks. With the rapid growth in such smart IoT networks and data rates, reliable transport layer protocols with efficient congestion control algorithms are required. The small Transmission Control Protocol/Internet Protocol (TCP/IP) stacks designed for IoT devices still lack efficient congestion control schemes. Multipath transmission control protocol (MPTCP) based congestion control algorithms are among the recent research topics. Many coupled and uncoupled congestion control algorithms have been proposed by researchers. The default congestion control algorithm for MPTCP is coupled congestion control by using the linked-increases algorithm (LIA). In battery powered smart meters, packet retransmissions consume extra power and low goodput results in poor system performance. In this study, we propose a modified Fast-Vegas-LIA hybrid congestion control algorithm (MFVL HCCA) for MPTCP by considering the requirements of a smart gas grid. Our novel algorithm operates in uncoupled congestion control mode as long as there is no shared bottleneck and switches to coupled congestion control mode otherwise. We have presented the details of our proposed model and compared the simulation results with the default coupled congestion control for MPTCP. Our proposed algorithm in uncoupled mode shows a decrease in packet loss up to 50% and increase in average goodput up to 30%.
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Cao, Yuanlong, Ruiwen Ji, Lejun Ji, Mengshuang Bao, Lei Tao, and Wei Yang. "Can Multipath TCP Be Robust to Cyber Attacks? A Measuring Study of MPTCP with Active Queue Management Algorithms." Security and Communication Networks 2021 (May 27, 2021): 1–11. http://dx.doi.org/10.1155/2021/9963829.
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With the development of social networks, more and more mobile social network devices have multiple interfaces. Multipath TCP (MPTCP), as an emerging transmission protocol, can fit multiple link bandwidths to improve data transmission performance and improve user experience quality. At the same time, due to the large-scale deployment and application of emerging technologies such as the Internet of Things and cloud computing, cyber attacks against MPTCP have gradually increased. More and more network security research studies point out that low-rate distributed denial of service (LDDoS) attacks are relatively popular and difficult to detect and are recognized as one of the most severe threats to network services. This article introduces six classic queue management algorithms: DropTail, RED, FRED, REM, BLUE, and FQ. In a multihomed network environment, we perform the performance evaluation of MPTCP under LDDoS attacks in terms of throughput, delay, and packet loss rate when using the six algorithms, respectively, by simulations. The results show that in an MPTCP-enabled multihomed network, different queue management algorithms have different throughput, delay, and packet loss rate performance when subjected to LDDoS attacks. Considering these three performance indicators comprehensively, the FRED algorithm has better performance. By adopting an effective active queue management (AQM) algorithm, the MPTCP transmission system can enhance its robustness capability, thus improving transmission performance. We suggest that when designing and improving the queue management algorithm, the antiattack performance of the algorithm should be considered: (1) it can adjust the traffic speed by optimizing the congestion control mechanism; (2) the fairness of different types of data streams sharing bandwidth is taken into consideration; and (3) it has the ability to adjust the parameters of the queue management algorithm in a timely and accurate manner.
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Shijer, Sameera Sadey, and Ahmad H. Sabry. "Analysis of performance parameters for wireless network using switching multiple access control method." Eastern-European Journal of Enterprise Technologies 4, no. 9(112) (August 31, 2021): 6–14. http://dx.doi.org/10.15587/1729-4061.2021.238457.
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The developments of wireless networks have directed to search for opportunities of a broad diversity of improved and new networking contributions. Wireless Asynchronous Transfer Mode (ATM) is a non-synchronous or random mode of transferring information. The advantages of circuit switching include dedicated connections and guaranteed traffic parameters and the benefits of packet switching are the efficiency at the physical layer and a more cost-effective design. ATM is the only protocol that offers the best of both communication methods. Although the Variable Bit-Rate (VBR) transmission presents a promising prospective of stable data quality, it is usually accompanied by network traffic overload and cell packet loss, which extensively weakens that potential. This work overcomes these concerns by developing a switching-based multiple access control model to improve the data transmission performance of wireless ATM. Therefore, this work discusses the effectiveness of the developed approach to minimize the cell packet losses and network traffic overload in wireless ATM. Three control access is processed; polling, token passing, and reservation algorithms for collision avoidance. The reservation stage reserves the data before sending, which includes two timeline intervals; a fixed-time reservation period, and variable data transmission interval. Using OPNET 10.5, the results show that the presented switching-based multiple access control model can achieve a throughput value of 98.3 %, data transmission delay of about 40.2 ms, and 0.024 % of packet losses during data transmission between the source and destination. It is demonstrated that the introduced method effectively transmits information without creating any network complexity and delay
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Khairi, Mutaz Hamed Hussien, Sharifah H. S. Ariffin, N. M. Abdul Latiff, and Kamaludin Mohamad Yusof. "Generation and collection of data for normal and conflicting flows in software defined network flow table." Indonesian Journal of Electrical Engineering and Computer Science 22, no. 1 (April 1, 2021): 307. http://dx.doi.org/10.11591/ijeecs.v22.i1.pp307-314.
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<a name="_Hlk31039004"></a><span lang="EN-US">In terms of network simplification and regulation, Software Defined Networking (SDN) is a new form of infrastructure that offers greater adaptability and flexibility. SDN, however, is an invention that is logically centralized. </span><span>In addition, the optimization of the control plane and data plane in SDN has become an area deserving of more attention. The flow in OpenFlow has been one of the essential parameters in the SDN standards, in which every individual flow includes packet matching fields, flow priority, separate counters, instructions for packet forwarding, flow timeouts and a cookie. This research work is conducted in order to produce and collect flows from the OpenFlow switch in two scenarios; in normal flows and when conflict policy rules are enforced in the network. In this article, the throughput is required to review and evaluate the conflict impact on two protocols as a performance metric; the Transmission Control Protocol (TCP) and the User Datagram Protocol (UDP) that flows via a forwarded plane. During the simulation of the SDN OpenFlow network, the metrics are tested using MININET. The results demonstrate that the existence of SDN conflict rules allows TCP and UDP to have a significant average change in bandwidth that eventually affects the network and operations performance.</span>
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ARYANTA, DWI, ARSYAD RAMADHAN DARLIS, and ARDHIANSYAH PRATAMA. "Implementasi Sistem IP PBX menggunakan Briker." ELKOMIKA: Jurnal Teknik Energi Elektrik, Teknik Telekomunikasi, & Teknik Elektronika 1, no. 2 (July 1, 2013): 117. http://dx.doi.org/10.26760/elkomika.v1i2.117.
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ABSTRAKVoIP (Voice over Internet Protocol) adalah komunikasi suara jarak jauh yang digunakan melalui jaringan IP. Pada penelitian ini dirancang sistem IP PBX dengan menggunakan teknologi berbasis VoIP. IP PBX adalah perangkat switching komunikasi telepon dan data berbasis teknologi Internet Protocol (IP) yang mengendalikan ekstension telepon analog maupun ekstension IP Phone. Software VirtualBox digunakan dengan tujuan agar lebih memudahkan dalam sistem pengoperasian Linux yang dimana program untuk membuat IP PBX adalah menggunakan Briker yang bekerja pada Operating System Linux 2.6. Setelah proses penginstalan Briker pada Virtualbox dilakukan implementasi jaringan IP PBX. Setelah mengimplementasikan jaringan IP PBX sesuai dengan topologi, kemudian melakukan pengujian success call rate dan analisis Quality of Service (QoS). Pengukuran QoS menggunakan parameter jitter, delay, dan packet loss yang dihasilkan dalam sistem IP PBX ini. Nilai jitter sesama user Briker (baik pada smartphone maupun komputer) mempunyai rata-rata berada pada nilai 16,77 ms. Sedangkan nilai packetloss yang didapat pada saat terdapat pada saat user 1 sebagai pemanggil telepon adalah 0%. Sedangkan persentase packet loss pada saat user 1 sebagai penerima telepon adalah 0,01%. Nilai delay pada saat berkomunikasi antar user berada pada 11,75 ms. Secara keseluruhan nilai yang didapatkan melalui penelitian ini, dimana hasil pengujian parameter-parameter QOS sesuai dengan standar yang telah direkomendasikan oleh ITU dan didapatkan nilai QoS dengan hasil “baik”.Kata Kunci: Briker, VoIP, QoS, IP PBX, Smartphone.ABSTRACTVoIP (Voice over Internet Protocol) is a long-distance voice communications over IP networks are used. In this study, IP PBX systems designed using VoIP -based technologies. IP PBX is a telephone switching device and data communication technology-based Internet Protocol (IP) which controls the analog phone extensions and IP Phone extensions. VirtualBox software is used in order to make it easier for the Linux operating system to create a program which is using briker IP PBX that works on Linux 2.6 Operating System. After the installation process is done briker on Virtualbox IP PBX network implementation. After implementing the IP PBX network according to the topology, and then do a test call success rate and analysis of Quality of Service (QoS). Measurement of QoS parameters using jitter, delay, and packet loss resulting in the IP PBX system. Jitter value briker fellow users (either on a smartphone or computer) has been on the average value of 16.77 ms. While the values obtained packetloss when there is 1 user when a phone caller is 0%. While the percentage of packet loss at user 1 as a telephone receiver is 0.01%. Delay value when communicating between users located at 11.75 ms. Overall value obtained through this study , where the results of testing the QOS parameters in accordance with the standards recommended by the ITU and the QoS values obtained with the results "good".Keywords: Briker, VoIP, QoS, IP PBX, Smartphone.
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Gao, Zhenyu, Jian Cao, Wei Wang, Huayun Zhang, and Zengrong Xu. "Online-Semisupervised Neural Anomaly Detector to Identify MQTT-Based Attacks in Real Time." Security and Communication Networks 2021 (September 13, 2021): 1–11. http://dx.doi.org/10.1155/2021/4587862.
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Industry 4.0 focuses on continuous interconnection services, allowing for the continuous and uninterrupted exchange of signals or information between related parties. The application of messaging protocols for transferring data to remote locations must meet specific specifications such as asynchronous communication, compact messaging, operating in conditions of unstable connection of the transmission line of data, limited network bandwidth operation, support multilevel Quality of Service (QoS), and easy integration of new devices. The Message Queue Telemetry Transport (MQTT) protocol is used in software applications that require asynchronous communication. It is a light and simplified protocol based on publish-subscribe messaging and is placed functionally over the TCP/IP protocol. It is designed to minimize the required communication bandwidth and system requirements increasing reliability and probability of successful message transmission, making it ideal for use in Machine-to-Machine (M2M) communication or networks where bandwidth is limited, delays are long, coverage is not reliable, and energy consumption should be as low as possible. Despite the fact that the advantage that MQTT offers its way of operating does not provide a serious level of security in how to achieve its interconnection, as it does not require protocol dependence on one intermediate third entity, the interface is dependent on each application. This paper presents an innovative real-time anomaly detection system to detect MQTT-based attacks in cyber-physical systems. This is an online-semisupervised learning neural system based on a small number of sampled patterns that identify crowd anomalies in the MQTT protocol related to specialized attacks to undermine cyber-physical systems.
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Danmanee, Tanapoom, Kulit Na Nakorn, and Kultida Rojviboonchai. "CU-MAC: A Duty-Cycle MAC Protocol for Internet of Things in Wireless Sensor Networks." ECTI Transactions on Electrical Engineering, Electronics, and Communications 16, no. 2 (April 9, 2018): 30–43. http://dx.doi.org/10.37936/ecti-eec.2018162.171332.
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Nowadays “Internet of Things” or IoT becomes the most popular technology in the Internet system. Types of devices and sensors have been connected as a network of devices and sensors. While a wireless sensor network is a traditional network of sensors that can be considered as a beginning point of IoT systems. Currently, these sensor data are not only exchanged within a local network but also are delivered to other devices in the Internet. Consequently, well-known organizations such as IEEE, IETF, ITU-T and ISO/IET are trying to set standards for wireless sensor devices in IoT systems. The recommended standard utilizes many of internet stack standards such as CoAP, UDP and IP. However, the traditional design of WSNs is to avoid using internet protocol in the system to reduce transmission overhead and power consumption due to resource limitation. Fortunately, the current technology in both hardware and software allow the internet standard to sufficiently operate in a small sensor. In this paper, we propose a MAC protocol named CU-MAC to efficiently support IoT standard that need request-respond communication or bi-direction communication. CU-MAC uses multi-channel communication to perform continuous and bi-directional data transfer at low duty-cycle. It also has a mechanism to overcome the hidden terminal problem. We evaluated the performance of CU-MAC on both simulation and real testbed based on Contiki OS. The result shows that CU-MAC outperforms other existing MAC protocols in term of packet delivery ratio at 98.7% and requires lower duty-cycle than others to operate in the high traffic environment.
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Pan, Wansu, Haibo Tan, Xiru Li, and Xiaofeng Li. "Improved RTT Fairness of BBR Congestion Control Algorithm Based on Adaptive Congestion Window." Electronics 10, no. 5 (March 6, 2021): 615. http://dx.doi.org/10.3390/electronics10050615.
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To alleviate the lower performance of Transmission Control Protocol (TCP) congestion control over complex network, especially the high latency and packet loss scenario, Google proposed the Bottleneck Bandwidth and Round-trip propagation time (BBR) congestion control algorithm. In contrast with other TCP congestion control algorithms, BBR adjusted transfer data by maximizing delivery rate and minimizing delay. However, some evaluation experiments have shown that the persistent queues formation and retransmissions in the bottleneck can lead to serious fairness issues between BBR flows with different round-trip times (RTTs). They pointed out that small RTT differences cause unfairness in the throughput of BBR flows and flows with longer RTT can obtain higher bandwidth when competing with the shorter RTT flows. In order to solve this fairness problem, an adaptive congestion window of BBR is proposed, which adjusts the congestion window gain of each BBR flow in network load. The proposed algorithms alleviate the RTT fairness issue by controlling the upper limit of congestion window according to the delivery rate and queue status. In the Network Simulator 3 (NS3) simulation experiment, it shows that the adaptive congestion window of BBR (BBR-ACW) congestion control algorithm improves the fairness by more than 50% and reduces the queuing delay by 54%, compared with that of the original BBR in different buffer sizes.
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khaing, Ei Ei, Mya Thet Khaing, Akari Myint Soe, and Shwe Sin Myat Than. "Implementation of Network Address Translation Using TCP/IP Model In Internet Communication System." International Journal of Scientific Research in Science, Engineering and Technology, June 6, 2020, 447–53. http://dx.doi.org/10.32628/ijsrset207385.
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Nowadays, many people will be used internet that for their work, communication, education, economic and organization necessary that is used today. Network address translation (NAT) is a method of remapping an IP address space into another by modifying network address information in the IP header of packets while they are in transit across a traffic routing device. A network is a system of hardware and software, put together for the purpose of communication and resource sharing. A network includes transmission hardware devise to interconnect transmission media and to control transmissions and software to decode and format data. The Internet protocol suite is the computer networking model and set of communications protocols used on the Internet and similar computer networks. Knowledge on how the internet is able to communicate with internet users is a mystery to some people. Internet communication need to be TCP/IP protocol which means that TCP is Transmission Control Protocol, or what is sometimes simply used to refer to Internet Protocol, is the basic unit for communication on the internet. This can also be applied to private internet, like Ethernet and so on. Despite TCP and IP being used interchangeably, there is a slight difference between the two in relation to the roles they play IP is directly responsible for obtaining internet addresses and then it is the work of TCP to deliver the data obtained to the addresses achieved by IP. TCP/IP provides end-to-end connectivity specifying how data should be packetized, addressed, transmitted, routed and received at the destination. This paper aim is described operation and models of TCP-IP suite in data communication network.
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"Implementation Based Approach to analyze MPLS and Segment Routing Traffic by Using ODL SDN Controller." International Journal of Advanced Trends in Computer Science and Engineering 10, no. 3 (June 15, 2021): 2324–36. http://dx.doi.org/10.30534/ijatcse/2021/1171032021.
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Network infrastructure in the data center is getting more complicated due to heavy network traffic, by increasing a network load leads high storage consumption and high deployment cost. We have proposed an effective traffic management mechanism by combining Segment Routing and Multipath Transmission Control Protocol TCP in a logically centralized controller environment. Segment Routing and Multipath TCP is used to define a logically centralized and physically distribution for Software Defined Network SDN. Segment Routing SR technology is used for simplifying Multi-protocol Label Switching MPLS and software- defined network SDN networks. Segment Routing makes it easy to assign a dynamic traffic according to customer’s requirements through the high-level application. This work presents a comprehensive comparison between MPLS and Segment Routing Traffic Engineering, both technologies are implemented in an SDN network environment, based on GNS3 network Emulator as data network Layer, Open-Daylight as SDN Controller, and Python scripts for sending Segment IDs as a part of network Orchestrator in Application Layer.
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Prayogi Wicaksana, Febri Hadi, and Aulia Fitrul Hadi. "Perancangan Implementasi VPN Server Menggunakan Protokol L2TP dan IPSec Sebagai Keamanan Jaringan." Jurnal KomtekInfo, August 14, 2021, 169–75. http://dx.doi.org/10.35134/komtekinfo.v8i3.128.
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One of the ways to maintain and improve the quality of service and security on the network of an agency of the Barangin Sub-District Office, Sawahlunto City (Central) is to add a VPN feature. Administrators who always monitor the flow of traffic by accessing routers and access points to find out network conditions. There are times when the Administrator is on a public network, he or she cannot access routers and access point devices because the Public IP obtained is Dynamic (random). To solve this problem, it is done through the Network Development Life Cycle (NDLC) method by combining the L2TP and IPsec VPN protocol systems on Mikrotik. A Virtual Private Network (VPN) is a private and secure network using a public network such as the internet. One of the bases for securing VPN technology is Internet Protocol Security (IPSec). IPSec is a protocol used to secure datagram transmission on TCP/IP-based networks. This study aims to design and implement a VPN network system by utilizing a public network, where this system provides advanced security enhancements on the internet network using IPSec. The information/data sent will be confidential with an automatic encryption method through the L2TP tunnel method from the server to the branch/client computer and vice versa. The VPN is implemented using a layer 2 (L2TP) tunneling protocol using two Mikrotik routers. There are few changes to the computer network configuration to minimize costs and implementation time. Tests are carried out to implement security on the network using the command prompt, where the admin observes packet loss and delay parameters to determine the increase in security quality on the network.
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Shim, Kyu-Seok, Ilkwon Sohn, Eunjoo Lee, Woojin Seok, and Wonhyuk Lee. "Enhance the ICS Network Security Using the Whitelist-based Network Monitoring Through Protocol Analysis." Journal of Web Engineering, February 17, 2021. http://dx.doi.org/10.13052/jwe1540-9589.2011.
Full textAbstract:
In our present technological age, most manual and semi-automated tasks are being automated for efficient productivity or convenience. In particular, industrial sites are rapidly being automated to increase productivity and improve work efficiency. However, while networks are increasingly deployed as an integral part of the automation of industrial processes, there are also many resultant dangers such as security threats, malfunctions, and interruption of industrial processes. In particular, while the security of business networks is reinforced and their information is not easily accessible, intruders are now targeting industrial networks whose security is relatively poor, wherein attacks could directly lead to physical damage. Therefore, numerous studies have been conducted to counter security threats through network traffic monitoring, and to minimize physical loss through the detection of malfunctions. In the case of industrial processes, such as in nuclear facilities and petroleum facilities, thorough monitoring is required as security issues can lead to significant danger to humans and damage to property. Most network traffic in industrial facilities uses proprietary protocols for efficient data transmission, and these protocols are kept confidential because of intellectual property and security reasons. Protocol reverse engineering is a preparatory step to monitor network traffic and achieve more accurate traffic analysis. The field extraction method proposed in this study is a method for identifying the structure of proprietary protocols used in industrial sites. From the extracted fields, the structure of commands and protocols used in the industrial environment can be derived. To evaluate the feasibility of the proposed concept, an experiment was conducted using the Modbus/TCP protocol and Ethernet/IP protocol used in actual industrial sites, and an additional experiment was conducted to examine the results of the analysis of conventional protocols using the file transfer protocol.