Bandwidth Requirements Of Users Research Articles

Analisis Quality of Service (QoS) Jaringan Internet Universitas Menggunakan Metode Hierarchical Token Bucket (HTB)

Qamarul Huda Badaruddin University is a private university located in Central Lombok Regency with an increasing number of students. Along with the increasing number of students, of course, the number of internet users on campus also increases, making traffic on the network more complex and requiring network management. Quality of Service (QoS) can be regarded as a terminology used to define the characteristics of a network service (Service) to find out how good the quality is in that service. With Quality of Service (QoS), bandwidth can be used optimally so as to improve the quality of internet services received by users. The application of appropriate bandwidth management needs to be done, one of which is using the HTB method, namely bandwidth management with bandwidth borrowing queuing techniques between classes or users. Which can allocate bandwidth according to the user's bandwidth requirements in accordance with the specified priority. The purpose of this study is to determine the Quality of Service (QoS) produced by the HTB method in conducting bandwidth management so as to improve existing internet services at Qamarul Huda Badaruddin University. The results of the Quality of Service (QoS) analysis based on the tests that have been carried out show that by applying the Hierarchical Token Bucket (HTB) method, satisfactory results are obtained, namely a value of 3.67. These results are obtained from the average test results on 4 parameters, namely delay, jitter, packet loss, and throughput.

Genetic Algorithms Optimized Adaptive Wireless Network Deployment

Advancements in UAVs have enabled them to act as flying access points that can be positioned to create an interconnected wireless network in complex environments. The primary aim of such networks is to provide bandwidth coverage to users on the ground in case of an emergency or natural disaster when existing network infrastructure is unavailable. However, optimal UAV placement for creating an ad hoc wireless network is an NP-hard and challenging problem because of the UAV’s communication range, unknown users’ distribution, and differing user bandwidth requirements. Many techniques have been presented in the literature for wireless mesh network deployment, but they lack either generalizability (with different users’ distributions) or real-time adaptability as per users’ requirements. This paper addresses the UAV placement and control problem, where a set of genetic-algorithm-optimized potential fields guide UAVs for creating long-lived ad hoc wireless networks that find all users in a given area of interest (AOI) and serve their bandwidth requirements. The performance of networks deployed using the proposed algorithm was compared with the current state of the art on several experimental simulation scenarios with different levels of communication among UAVs, and the results show that, on average, the proposed algorithm outperforms the state of the art by 5.62% to 121.73%.

Peer–Peer Communication Using Novel Slice Handover Algorithm for 5G Wireless Networks

The goal of 5G wireless networks is to address the growing need for network services among users. User equipment has progressed to the point where users now expect diverse services from the network. The latency, reliability, and bandwidth requirements of users can all be classified. To fulfil the different needs of users in an economical manner, while guaranteeing network resources are resourcefully assigned to consumers, 5G systems plan to leverage technologies like Software Defined Networks, Network Function Virtualization, and Network Slicing. For the purpose of ensuring continuous handover among network slices, while catering to the advent of varied 5G application scenarios, new mobility management techniques must be adopted in Sliced 5G networks. Users want to travel from one region of coverage to another region without any fading in their network connection. Different network slices can coexist in 5G networks, with every slice offering services customized to various QoS demands. As a result, when customers travel from one region of coverage to another, the call can be transferred to a slice that caters to similar or slightly different requirements. The goal of this study was to develop an intra- and inter-slice algorithm for determining handover decisions in sliced 5G networks and to assess performance by comparing intra- and inter-slice handovers. The proposed work shows that an inter-slice handover algorithm offers superior quality of service when compared to an intra-slice algorithm.

Secure and Efficient Data Deduplication in JointCloud Storage

Data deduplication can efficiently eliminate data redundancies in cloud storage and reduce the bandwidth requirement of users. However, most previous schemes depending on the help of a trusted key server (KS) are vulnerable and limited because they suffer from revealing information, poor resistance to attacks, great computational overhead, etc. In particular, if the trusted KS fails, the whole system stops working, i.e., single-point-of-failure. In this paper, we propose a Secure and Efficient data Deduplication scheme (named SED) in a JointCloud storage system which provides the global services via collaboration with various clouds. SED also supports dynamic data update and sharing without the help of the trusted KS. Moreover, SED can overcome the single-point-of-failure that commonly occurs in the classic cloud storage system. According to the theoretical analyses, our SED ensures the semantic security in the random oracle model and has strong anti-attack ability such as the brute-force attack resistance and the collusion attack resistance. Besides, SED can effectively eliminate data redundancies with low computational complexity and communication and storage overhead. The efficiency and functionality of SED improves the usability in client-side. Finally, the comparing results show that the performance of our scheme is superior to that of the existing schemes.

ANALISIS QUALITY OF SERVICE (QOS) PADA JARINGAN INTERNET DI UNIVERSITAS BINA INSAN LUBUKLINGGAU MENGGUNAKAN METODE HIERARCHICAL TOKEN BUCKET (HTB)

Abstract - One problem that is often faced by internet users is slow internet access. Specifically, when there are many users sharing internet bandwidth. This problem can occur because no bandwidth management system is used. Therefore it is necessary to apply the appropriate bandwidth management, one of which is HTB. Bandwidth management with queuing bandwidth lending techniques between classes or users. Which can allocate bandwidth according to user bandwidth requirements according to specified priorities. This study uses data collection methods, by observing and recording directly at the research site (observation), conducting question and answer directly to the source (Interview), and documentation by reading literary books. The results obtained from the measurement of QoS parameters using the HTB method on the Bina Insan University network, it can produce an average QoS index value of 3, and included in the category is good. It can be concluded that the HTB method is feasible to be applied at UNIBI in performing bandwidth management as evidenced by the 3 (three) QoS index value and has a good category. Keywords: Quality Of Service, Hierarchical Token Bucket, Computer Network

Wireless Indoor Resource Optimization Meeting Prescribed User Capacity Requirements

The optimal locations of transmitters in a building, that best meet the bandwidth requirements of all users, was studied in the past by a small number of authors. They used empirical path loss models to constrain an optimization cost function. This paper concentrates on using a signal-to-interference ratio cost function with realistic field parameters obtained using a fast ray-tracing algorithm. The solution obtained is shown to perform well in realistic scenarios, meeting a large proportion of user bandwidth requirements. However, it does not perform well in an oversaturated environment.

MPTCP Tunnel: An Architecture for Aggregating Bandwidth of Heterogeneous Access Networks

Fixed and cellular networks are two typical access networks provided by operators. Fixed access network is widely employed; nevertheless, its bandwidth is sometimes not sufficient enough to meet user bandwidth requirements. Meanwhile, cellular access network owns unique advantages of wider coverage, faster increasing link speed, more flexible deployment, and so forth. Therefore, it is attractive for operators to mitigate the bandwidth shortage by bundling these two. Actually, there have been existing schemes proposed to aggregate the bandwidth of two access networks, whereas they all have their own problems, like packet reordering or extra latency overhead. To address this problem, we design new architecture, MPTCP Tunnel, to aggregate the bandwidth of multiple heterogeneous access networks from the perspective of operators. MPTCP Tunnel uses MPTCP, which solves the reordering problem essentially, to bundle multiple access networks. Besides, MPTCP Tunnel sets up only one MPTCP connection at play which adapts itself to multiple traffic types and TCP flows. Furthermore, MPTCP Tunnel forwards intact IP packets through access networks, maintaining the end-to-end TCP semantics. Experimental results manifest that MPTCP Tunnel can efficiently aggregate the bandwidth of multiple access networks and is more adaptable to the increasing heterogeneity of access networks than existing mechanisms.

Analysis of Bandwidth Requirement of Users in Flexible Reuse Cellular Networks

In flexible frequency reuse scheme namely fractional frequency reuse (FFR), the total frequency resource is divided logically into cell center and cell edge user bands. However, while frequency resource is partitioned into two segments, the amount of average bandwidth required by cell center and cell edge users is quite different and influences the system performance and design. We developed the analytical approach to evaluate the average bandwidth required by cell center and cell edge users for real time (RT) traffic. It is seen that the analytical results are closely matched with the simulations and indicate that the model is a good approximation for bandwidth requirement of users in a cellular environment.

Optimal Bandwidth Assignment for Multiple-Description-Coded Video

In video streaming over multicast network, user bandwidth requirement is often heterogeneous possibly with orders of magnitude difference (say, from hundreds of kb/s for mobile devices to tens of Mb/s for high-definition TV). Multiple description coding (MDC) can be used to address this bandwidth heterogeneity issue. In MDC, the video source is encoded into multiple independent descriptions. A receiver, depending on its available bandwidth, joins different descriptions to meet their bandwidth requirements. An important but challenging problem for MDC video multicast is how to assign bandwidth to each description in order to maximize overall user satisfaction. In this paper, we investigate this issue by formulating it as an optimization problem, with the objective to maximize user bandwidth experience by taking into account the encoding inefficiency due to MDC. We prove that the optimization problem is NP-hard. However, if the description number is larger than or equal to a certain threshold (e.g., if the minimum and maximum bandwidth requirements are 100 kb/s and 10 Mb/s, respectively, such threshold is seven descriptions), there is an exact and simple solution to achieve maximum user satisfaction, i.e., meeting all the bandwidth requirements. For the case when the description number is smaller, we present an efficient heuristic called simulated annealing for MDC bandwidth assignment (SAMBA) to assign bandwidth to each description given the distribution of user bandwidth requirement. We evaluate our algorithm using simulations. SAMBA achieves virtually the same optimal performance based on exhaustive search. By comparing with other assignment algorithms, SAMBA significantly improves user satisfaction. We also show that, if the coding efficiency decreases with the number of descriptions, there is an optimal description number to achieve maximal user satisfaction.

Multipath Routing for Video Delivery Over Bandwidth-Limited Networks

The delivery of quality video service often requires high bandwidth with low delay or cost in network transmission. Current routing protocols such as those used in the Internet are mainly based on the single-path approach (e.g., the shortest-path routing). This approach cannot meet the end-to-end bandwidth requirement when the video is streamed over bandwidth-limited networks. In order to overcome this limitation, we propose multipath routing, where the video takes multiple paths to reach its destination(s), thereby increasing the aggregate throughput. We consider both unicast (point-to-point) and multicast scenarios. For unicast, we present an efficient multipath heuristic (of complexity O(|V|/sup 3/)), which achieves high bandwidth with low delay. Given a set of path lengths, we then present and prove a simple data scheduling algorithm as implemented at the server, which achieves the theoretical minimum end-to-end delay. For a network with unit-capacity links, the algorithm, when combined with disjoint-path routing, offers an exact and efficient solution to meet a bandwidth requirement with minimum delay. For multicast, we study the construction of multiple trees for layered video to satisfy the user bandwidth requirements. We propose two efficient heuristics on how such trees can be constructed so as to minimize the cost of their aggregation subject to a delay constraint.

Protocol design for scalable and reliable group rekeying

We present the design and specification of a protocol for scalable and reliable group rekeying together with performance evaluation results. The protocol is based upon the use of key trees for secure groups and periodic batch rekeying. At the beginning of each rekey interval, the key server sends a rekey message to all users consisting of encrypted new keys (encryptions, in short) carried in a sequence of packets. We present a scheme for identifying keys, encryptions, and users, and a key assignment algorithm that ensures that the encryptions needed by a user are in the same packet. Our protocol provides reliable delivery of new keys to all users eventually. It also attempts to deliver new keys to all users with a high probability by the end of the rekey interval. For each rekey message, the protocol runs in two steps: a multicast step followed by a unicast step. Proactive forward error correction (FEC) multicast is used to reduce delivery latency. Our experiments show that a small FEC block size can be used to reduce encoding time at the server without increasing server bandwidth overhead. Early transition to unicast, after at most two multicast rounds, further reduces the worst-case delivery latency as well as user bandwidth requirement. The key server adaptively adjusts the proactivity factor based upon past feedback information; our experiments show that the number of NACKs after a multicast round can be effectively controlled around a target number. Throughout the protocol design, we strive to minimize processing and bandwidth requirements for both the key server and users.

An efficient scheme to achieve weighted max-min fairness on bandwidth allocation

In this paper, a scheme is proposed to achieve weighted max-min fairness on bandwidth allocation. The scheme supports various bandwidth requirements of users according to their weights. With the reservation strategy, the minimum bandwidth of each flow is guaranteed. The scheme achieves fair allocation rapidly when the network condition changes. The basic idea is to find the bandwidth share of each active flow at its bottleneck link and then assign this bottleneck share as its fair allocation. The correctness of the scheme is also proven. The scheme is efficient since it takes O(log N ) time to allocate bandwidth to N active flows. Simulations performed validate that the scheme reacts quickly to the changes of network condition and re-establishes the state of fair allocation fast.

A Nonstationary Offered-Load Model for Packet Networks

Motivated by the desire to appropriately account for complex features of network traffic revealed in traffic measurements, such as heavy-tail probability distributions, long-range dependence, self similarity and nonstationarity, we propose a nonstationary offered-load model. Connections of multiple types arrive according to independent nonhomogeneous Poisson processes, and general bandwidth stochastic processes (not necessarily Markovian) describe the individual user bandwidth requirements at multiple links of a communication network during their connections. We obtain expressions for the moment generating function, mean and variance of the total required bandwidth of all customers on each link at any designated time. We justify Gaussian approximations by establishing a central limit theorem for the offered-load process. We also obtain a Gaussian approximation for the time-dependent buffer-content distribution in an infinite-capacity buffer with constant processing rate. The offered-load model can be used for predicting future bandwidth requirements; we then advocate exploiting information about the history of connections in progress.