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One of the most important developments in the energy industry is the evolution of smart grids, which record minute details of voltage levels, energy usage, and other critical electrical variables through General Packet Radio Service (GPRS)-enabled meters. This phenomenon creates an extensive dataset for the optimization of the grid system. However, the minute-by-minute energy details recorded by GPRS meters are challenging to store and manage in physical storage resources (old techniques lead to a memory shortage). This study investigates using the distributed file system, replicated Gluster, as a reliable storage option for handling and protecting the enormous volumes of data produced by smart grid components. This study performs two essential tasks. (1) The storage of virtual data received from GPRS meters and load flow analysis of SynerGee Electric 4.0 software from the smart grid (we have extracted electrical data from 16 outgoing feeders, distributed lines, in this manuscript). (2) Tele-trafficking is performed to check the performance of replicated Gluster (RG) for virtual data (electrical data received from the smart grid) storage in terms of User Datagram Protocol (UDP), Transmission Control Protocol (TCP), data flow, and jitter delays. This storage technique provides more opportuni11ty to analyze and perform smart techniques efficiently for future requirement, analysis, and load estimation in smart grids compared to traditional storage methods. [ABSTRACT FROM AUTHOR]

Recently, many new network paths have been introduced while old paths are still in use. The trade-offs remain vague and should be further addressed. Since last decade, the Internet is playing a major role in people's lives, and the demand on the Internet in all fields has increased rapidly. In order to get a fast and secure connection to the Internet, the networks providing the service should get faster and more reliable. Many network data paths have been proposed in order to achieve the previous objectives since the 1970s. It started with the Transmission Control Protocol (TCP) and the User Datagram Protocol (UDP) and later followed by several more modern paths including Quick UDP Internet Connections (QUIC), remote direct memory access (RDMA), and the Data Plane Development Kit (DPDK). This raised the question on which data path should be adopted and based on which features. In this work, we try to answer this question using different perspectives such as the protocol techniques, latency and congestion control, head of line blocking, the achieved throughput, middleboxes consideration, loss recovery mechanisms, developer productivity, host resources utilization and targeted application. [ABSTRACT FROM AUTHOR]

The current User Datagram Protocol (UDP) causes unfairness and bufferbloats to delay sensitive applications due to the uncontrolled congestion and monopolization of available bandwidth.This causes call drops and frequent communication/connection loss in delay sensitive applications such as VoIP. We present a Responsive Control Protocol using Bottleneck Bandwidth and Round trip propagation time (RCP-BBR) as an alternate solution to UDP. RCP-BBR achieves low latency, high throughput, and low call drops ratio by efficiently customizing Transmission Control Protocol (TCP) Bottleneck Bandwidth and Round-trip propagation time (TCP-BBR) congestion control. We conducted comprehensive experiments, and the results show that proposed protocol achieves better throughput over UDP in stable networks. Moreover, in unstable and long-distanced networks, RCP-BBR achieved smaller queues in deep buffers and lower delays as compared to UDP, which performed poorly by keeping delays above the call drop threshold. [ABSTRACT FROM AUTHOR]

IoT wearable devices that leverage the LPWAN (Low Power Wide Area Network) connectivity, in particular the NB-IoT (Narrow Band IoT) technology, commonly rely on UDP (User Datagram Protocol) packets to communicate with the endpoint to collect data. Its advantage is that of minimal bandwidth usage but its drawback is the lack of acknowledgements by the endpoint. One option is to use TCP (Transmission Control Protocol) or implement a custom acknowledgement protocol at a superior level in the network stack. This paper presents the implementation and testing of a simple and lightweight software synchronization layer on top of the UDP protocol, for the transmission of live EKG signal data, with verification of data integrity and the detection of sampling points loss. [ABSTRACT FROM AUTHOR]

Cloud computing, with its staggering and on-demand services had revamped the technology so far. Cloudconsumers are freely to use the applications and software on the premises of Pay-as-you go concept. This concept decreased the cost and make the services less expensive and more reliable. One of the most important characteristic of cloud structure is on demand self-service. Cloud computing applications can be accessed anywhere at any time with much less cost. As cloud provide its consumers with its tremendous on demand services, besides this it is surviving from the excruciating security issues that are discourteous towards the cloud. There are, as many different attacks that results in making the servers down. One of the most hazardous attack is DDoS. This paper hiloghted the DDoS attack and its prevention technique which results in making the server side less vulnerable. The scenario includes, a transmission of million and trillion of packets in the form of DDoS at cloud-based websites, thus making it differentiated though different hosts. Making use of operating systems such as ParrotSec to make the attack possible. Last step includes detection and prevention through the most effective algorithms namely, Naïve Bayes and Random forest. This paper also focused the categories of attacks on cloud computing.

Flood attacks on a network occur when attackers send a very high volume of traffic to a system. This leads to an exhaustion of resources of the targeted system. Such a system or infrastructure under attack won't be able to cope up with the services it needs to provide due to capacity overload. These types of attacks are also called as Denial of Service (DoS) attacks. As it is essential for companies and organizations to provide high speed and high-quality services the network in which it operates needs to be secured against DoS attacks. As it is said prevention is better than cure, it is always better to detect an attack in the preliminary stages itself and deny an attacker any opportunity to continue the assault on the system. There are various kinds of DoS attacks, in this study the focus is on three kinds of attacks and its detection ICMP flood attack, UDP flood attack and TCP SYN flood attack. The detection mechanism employed is based on packet symmetry which can be used in to monitor the data flow in a network and detect any malicious anomalies in the network traffic at any instant. The experimental results, show that the packet symmetry is a reliable metric to detect attacks on the network by checking the data flow in the ICMP, UDP and TCP layers. [ABSTRACT FROM AUTHOR]

Teleconferencing is the act of meeting and exchanging information through a telecommunication medium. In teleconferencing system, network engineers are faced with different challenges like best protocols requirements; impact of Quality of Service (QoS) to be used to meet up with the latest network protocols and appliances in the design of teleconference system. This paper presented a comparative analysis on two network protocols in a teleconference system, Transmission Control Protocol (TCP) versus a User Datagram Protocol (UDP) was evaluated and compared applying Optimized Network Engineering Tools (OPNET) simulator. The experimental result showed differences in number of videoconference sessions, with few Packet Per Seconds (PPS) recorded in both the UDP and TCP, thereby reflecting the correctness of this comparative evaluation task. [ABSTRACT FROM AUTHOR]

Cloud computing, with its staggering and on-demand services had revamped the technology so far. Cloud consumers are freely to use the applications and software on the premises of Pay-as-you go concept. This concept decreased the cost and make the services less expensive and more reliable. One of the most important characteristic of cloud structure is on demand self-service. Cloud computing applications can be accessed anywhere at any time with much less cost. As cloud provide its consumers with its tremendous on demand services, besides this it is surviving from the excruciating security issues that are discourteous towards the cloud. There are, as many different attacks that results in making the servers down. One of the most hazardous attack is DDoS. This paper hiloghted the DDoS attack and its prevention technique which results in making the server side less vulnerable. The scenario includes, a transmission of million and trillion of packets in the form of DDoS at cloud-based websites, thus making it differentiated though different hosts. Making use of operating systems such as ParrotSec to make the attack possible. Last step includes detection and prevention through the most effective algorithms namely, Naive Bayes and Random forest. This paper also focused the categories of attacks on cloud computing. [ABSTRACT FROM AUTHOR]

Survivability is an issue of great concern in high speed networks. Optical Packet Switching (OPS) is an established technique in the area of survivable high speed future networks. In this paper, network performance is evaluated in response to ingress packet size variation. The performance is evaluated for both protocols: Transmission Control Protocol (TCP) and User Datagram Protocol (UDP). Analysis is performed on mesh networks for different ingress packet size. Response of both transport layer protocols has been observed under single link failure. Network throughput with TCP is higher as compared to throughput with UDP. Network throughput is highest at 500 byte packet size, which is less than the configured Maximum Segment Size (MSS) of transport layer protocols. [ABSTRACT FROM AUTHOR]

Smart Home concept is to connect several devices to perform control and the device to be controlled to help any human needs. So it should have one device being Host and several devices being Client. The environment in this case is a home that there is too much technology that planted technologies that has much influence at human activities is called pervasive computing. The device communication should be has a protocol, at the research used UDP protocol because it doesn't need to negotiate before performing communication like a TCP. With UDP behavior did not use to validate at every send or deliver data because at Smart Home environment doesn't need a large or big data being processed. LabView used at this research based on previous research, but MyRIO has used this research to perform Client and Host performed by PC. So it can measure how usable the design that created. This research proves that LabView and MyRIO successfully implemented with the design that created to perform Pervasive Computing for Smart Home environment. Functional testing scenario conducted by every condition that on system environment, based on design all testing scenario successfully and working as well as expected. [ABSTRACT FROM AUTHOR]

LAN Chat Messenger using TCP/IP offers reliability, security and zero cost communication among staff members of a company. As well, this study offers file transfer. It helps to solve the communication problems that related to time and cost. The proposed protocol facilitates information exchange among individuals by providing many communication options. It is a standalone application using JAVA as the programming language and tested at LANs of our institute (college's Labs networks). [ABSTRACT FROM AUTHOR]

Fixed-point models have already been successfully used to analytically study networks consisting of persistent TCP flows only, or mixed TCP/UDP flows with a single queue per link and differentiated buffer management for these two types of flows. In the current study, we propose a nested fixed-point analytical method to obtain the throughput of persistent TCP and UDP flows in a network of routers supporting class-based weighted fair queuing allowing the use of separate queues for each class. In particular, we study the case of two classes where one of the classes uses drop-tail queue management and is intended for only UDP traffic. The other class targeting TCP, but also allowing UDP traffic for the purpose of generality, is assumed to employ active queue management. The effectiveness of the proposed analytical method is validated in terms of accuracy using ns-3 simulations and the required computational effort. [ABSTRACT FROM AUTHOR]

This paper analyzes latency in a High-Speed Packet Access network appropriately resourced for emulated Machine-to-Machine and Online Gaming traffic. Transmission Control Protocol (TCP) and User Datagram Protocol (UDP) are used for conveying application data and the measurement results from the end-user perspective are compared. In TCP case, traffic traces were recorded in two points of the core network as well (Gn interface and the firewall), and the overall delay is dissected to portions belonging to different parts of the network (access, core, backbone). The diversity of the traffic patterns used made it possible to draw conclusions concerning selectivity of certain parts of the network towards different traffic patterns, indicating the direction of future research on reducing latency for subject emerging application domains in legacy networks. [ABSTRACT FROM AUTHOR]

This paper suggest remote control technic using TCP/IP protocol network after UART serial communication implemented through RS-232 cable to ethernet module. This is multidisplinary signal technique which is PC signal is converted to internet communication signal. Ethernet module is WIZ1000SR. This is gate module which convert RS-232 protocol to TCP/IP protocol. This module convert serial data transmitted serial device to TCP/IP type data, in contrast TCP/IP data received through network converted to serial data transmitted to serial device. This paper propose t [ABSTRACT FROM AUTHOR]

Although most of the data traffic in Internet is HTTP-based, multimedia applications will soon dominate a large percentage of the traffic. These applications require satisfactory level of bandwidth as they normally have large datasets. Under the limitation of network bandwidth, multimedia traffic may become a source for congestion; particularly when UDP is used due to the absence of flow control or congestion control mechanisms in this protocol. Avoiding congestion can be done if arrival rate to gateways is maintained close to outgoing link capacity while maintain small queue lengths at the routers. This guarantees the availability of buffer capacity for successful buffering and consequent forwarding in case of temporary traffic upsurges which could otherwise cause buffer overflows and packet loss. This research work introduces a congestion-aware and friendly UDP-based multimedia system which can help to lessening congestion occurrence and improve network performance, principally in real-time environments. The evaluation results show that the proposed system helps in controlling congestion, reducing packet loss, increasing throughput, and providing improved network utilization. [ABSTRACT FROM AUTHOR]

Lately, the development and progress have become significant in the field of information technology, especially in the field of data transmission via internet. One kind of the data transfer is voice. This part of the development in the field of mobile devices, which is possibility supplying the device by internet service .These developments in the field of technology are concurring many problems, where it had a direct impact on this development and the turnout by the users. [ABSTRACT FROM AUTHOR]

This article presents the use of fluid flow approximation to model interactions between a set of TCP, UDP and XCP flows in the environment of IP routers using AQM (Active Queue Management) algorithms to control traffic congestion. In contrast to other works, independent UDP and TCP streams are considered and the model allows to start and end data transmissions in TCP, UDP and XCP streams at any time moment. It incorporates several Active Queue Management mechanisms: RED, NLRED, CHOKe. [ABSTRACT FROM AUTHOR]

Scanning of ports on a computer occurs frequently on the Internet. An attacker performs port scans of Internet protocol addresses to find vulnerable hosts to compromise. However, it is also useful for system administrators and other network defenders to detect port scans as possible preliminaries to more serious attacks. It is a very difficult task to recognize instances of malicious port scanning. In general, a port scan may be an instance of a scan by attackers or an instance of a scan by network defenders. In this survey, we present research and development trends in this area. Our presentation includes a discussion of common port scan attacks. We provide a comparison of port scan methods based on type, mode of detection, mechanism used for detection and other characteristics. This survey also reports on the available data sets and evaluation criteria for port scan detection approaches. [ABSTRACT FROM AUTHOR]

The IEEE 802.21 Media Independent Handover (MIH) working group is developing a set of mechanisms to facilitate migration of mobile users between access networks that use different link-layer technologies. Among these are mobility managers that create and process signaling messages to facilitate handovers. The MIH signaling architecture being developed in the Internet Engineering Task Force (IETF) allows any transport layer protocol to carry MIH messages. The IETF has considered using the unreliable but lightweight transport available with the User Datagram Protocol (UDP) as well as the reliable stream-oriented transport with congestion control offered by the Transmission Control Protocol (TCP). In this paper we develop mathematical models that result in expressions for the characteristic function of the time required to complete exchanges of an arbitrary number of MIH signaling messages between a mobile node (MN) and a remote mobility manager (MM). Our models also provide expressions for the average amount of overhead associated with MIH message exchanges due to retransmissions either by the MIH signaling entities or by the transport-layer protocol. In addition, we provide simulation results that confirm the results from the mathematical model and illustrate the effect of varying transport parameters such as the TCP maximum retransmission timeout. [ABSTRACT FROM AUTHOR]

In this paper we present our solution for improving the quality of communication in our particular desktop grid system, QADPZ, which consists of a dual communication mechanism. Thus, in order to improve the efficiency of communication, QADPZ support two different protocols (UDP and TCP/IP). To deal with the unreliability of UDP, an additional, more reliable level of communication has been added to the system, which works similarly with the real life postal service, delivering and receiving high-level message and implements a reliable, confirmation-based message exchange protocol. We point out that an unreliable transport mechanism that deals with packet loss gracefully and does not exhibit extreme losses can compete well with other more reliable transport protocols. [ABSTRACT FROM AUTHOR]

There is a vast literature on the throughput analysis of the IEEE 802.11 media access control (MAC) protocol. However, very little has been done on investigating the interplay between the cotlisioc avoidance mechanisms of the 802.11 MAC protocol and the dynamics of upper layer transpod protocols. In this paper, we tackle this issue from an analytical, simulative, and experimental perspective. Specifically, we develop Markov chain models to compute the distribution of the number of active stations in an 802.11 wireless local area network (WLAN) when long-lived Transmission Control Protocol (TCP) connections compete with finite-loed User Datagram Protocol (UDP) flows. By embedding these distributions in the MAC protocol modeling, we derive approximate but accurate expressions of the TCP and UDP throughput. We validate the model accuracy through performance tests carried out in a real WLAN for a wide range of configurations. Our analytical model and the supporting experimental outcomes show that 1) the total TCP throughput is basically independent of the number of open TCP connections and the aggregate TCP traffic can be equivalently modeled as two saturated flows and 2) in the saturated regime, n UDP flows obtain about n times the aggregate throughput achieved by the TCP flows, which is independent of the overall number of persistent TCP connections. [ABSTRACT FROM AUTHOR]

We propose a bandwidth control method to increase or decrease the entire assigned upstream bandwidth by monitoring the queue length in order to prevent a drop in the throughput and an increase in the delay time when congestion is produced by the traffic generated by multiple users in a passive optical network (PON)-type optical access system for economically providing wideband multimedia services. The ability of the proposed method to avoid congestion, improve the Transmission Control Protocol (TCP) throughput, reduce the delays, and assure fairness between the TCP and User Datagram Protocol (UDP) traffic is demonstrated in computer simulations by applying the proposed method to the Broadband-PON (B-PON) system which complies with the ITU-T standard. © 2007 Wiley Periodicals, Inc. Electron Comm Jpn Pt 2, 90(7): 41– 55, 2007; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjb.20374 [ABSTRACT FROM AUTHOR]

Wireless networks beyond 2G aim at supporting real-time applications such as VolP. Before a user can start a VolP session, the end-user terminal has to establish the session using signaling protocols such as H.323 and session initiation protocol (SIP) in order to negotiate media parameters. The time interval to perform the session setup is called the session setup time. It can be affected by the quality of the wireless link, measured in terms of frame error rate (FER), which can result in retransmissions of packets lost and can lengthen the session setup time. Therefore, such protocols should have a session setup time optimized against loss. One way to do so is by choosing the appropriate retransmission timer and the underlying protocols. In this paper, we focus on SIP session setup delay and propose optimizing it using an adaptive retransmission timer. We also evaluate SIP session setup performances with various underlying protocols (transport control protocol (TCP), user datagram protocol (UDP), radio link protocols (RLPs)) as a function of the FER. For 19.2 Kbps channel, the SIP session setup time can be up to 6.12s with UDP and 7s with TCP when the FER is up to 10 percent. The use of RLP (1,2, 3) and RLP (1, 1, 1, 1, 1, 1) puts the session setup time down to 3.4s under UDP and 4s under TCP for the same FER and the same channel bandwidth. We also compare SIP and H.323 performances using an adaptive retransmission timer: SIP outperforms H.323, especially for a FER higher than 2 percent. [ABSTRACT FROM AUTHOR]