Packet Delay Variation Research Articles

Investigating the Analysis of Power Saving Mode in IEEE 802.11 for Wi-Fi Communication

One of the major global issues today is energy consumption. Consequently, power management was introduced in various communication technologies. For IEEE 802.11wireless communication, there is a Power Saving Mode scheme (PSM) for increase the battery life of cell phone. In this PSM, there are two key parameters: beacon period interval (BI) and listen interval(LI). In most work these values are chosen arbitrary. Here, a scheme to determine the optimal BI and LI for accomplishing the most astounding conceivable vitality proficiency is introduced. This is implemented with the application of a numerical sample to the standard IEEE 802.11 PSM and Access Point-PSM (AP-PSM) schemes. To ensure the quality of network performance analysis on the normal and change of parcel delays is doing. The well-known queuing (M/G/I) model with bulk services are utilized. After the implementation of the proposed analysis, “maximum rest plan time ratio optimal Sleep Scheme (OSS)” which is when participate stations stay in the doze mode it can be determined. In this research shows that the optimal BI and LI produce optimal OSS time ratio scheme also achieved optimal average and variance of packet delay.

Cooperative Multi PTP Slaves for Timing Improvement in an fGn Environment

Papers dealing with the Ethernet network have shown that the packet delay variation (PDV) is a long range dependency process which may be modeled as a fractional Gaussian noise. In this letter, a closed-form approximated expression is derived for the offset variance valid for multiple precision time protocol (PTP) slaves (IEEE1588 v2) cooperating together in the network. Using this expression, we obtain, when there is long range dependence in the PDV, the total number of PTP slaves needed to receive the same offset variance as that obtained for the case of white noise PDV.

Channel Clustering and QoS Level Identification Scheme for Multi-Channel Cognitive Radio Networks

The increasing popularity of wireless services and devices necessitates high bandwidth requirements; however, spectrum resources are not only limited but also heavily underutilized. Multiple license channels that support the same levels of QoS are desirable to resolve the problems posed by the scarcity and inefficient use of spectrum resources in multi-channel cognitive radio networks (MCRNs). One reason is that multimedia services and applications have distinct, stringent QoS requirements. However, due to a lack of coordination between primary and secondary users, identifying the QoS levels supported over available licensed channels has proven to be problematic and has yet to be attempted. This article presents a novel Bayesian non-parametric channel clustering scheme, which identifies the QoS levels supported over available license channels. The proposed scheme employs the infinite Gaussian mixture model and collapsed Gibbs sampler to identify the QoS levels from the feature space of the bit rate, packet delivery ratio, and packet delay variation of licensed channels. Moreover, the real measurements of wireless data traces and comparisons with baseline clustering schemes are used to evaluate the performance of the proposed scheme.

IoT Traffic Management and Integration in the QoS Supported Network

This paper proposes: 1) a traffic flow management policy, which allocates and organizes machine type communication (MTC) traffic flows network resources sharing within evolved packet system (EPS); 2) an access element as a wireless sensor network gateway for providing an overlaying access channel between the machine type devices and EPS; and 3) it addresses the effect and interaction in the heterogeneity of applications, services and terminal devices, and the related quality of service (QoS) issues among them. This paper overcomes the problems of network resource starvation by preventing deterioration of network performance. The scheme is validated through simulation, which indicates the proposed traffic flow management policy outperforms the current traffic management policy. Specifically, simulation results show that the proposed model achieves an enhancement in QoS performance for the MTC traffic flows, including a decrease of 99.45% in packet loss rate (PLR), a decrease of 99.89% in packet end to end (E2E) delay, a decrease of 99.21% in packet delay variation (PDV). Furthermore, it retains the perceived quality of experience of the real time application users within high satisfaction levels, such as the voice over long term evolution service possessing a mean opinion score (MOS) of 4.349 and enhancing the QoS of a video conference service within the standardized values of a 3GPP body, with a decrease of 85.28% in PLR, a decrease of 85% in packet E2E delay and a decrease of 88.5% in PDV.

Performance Evaluation of RIPng, EIGRPv6 and OSPFv3 for Real Time Applications

In this modern Internet era and the transition to IPv6, routing protocols must adjust to assist this transformation. RIPng, EIGRPv6 and OSPFv3 are the dominant IPv6 IGRP (Interior Gateway Routing Protocols). Selecting the best routing protocol among the available is a critical task, which depends upon the network requirement and performance parameters of different real time applications. The primary motivation of this paper is to estimate the performance of these protocols in real time applications. The evaluation is based on a number of criteria including: network convergence duration, Http Page Response Time, DB Query Response Time, IPv6 traffic dropped, video packet delay variation and video packet end to end delay. After examining the simulation results, a conclusion will be extracted to reveal the findings of which protocol performs the best upon implementation within a IPv6 WAN. OPNET modeler simulator is used to evaluate the accomplishment of these protocols. To get the results, three scenarios are designed, one for each protocol.

Ring Optical Packet switched (OPS) network: Quality of Service (QoS) and traffic model

Optical Packet Switched (OPS) technology is under study as a promising solution for addressing the rapid growth of network-intensive applications having different Quality of Service (QoS) requirements. By employing Optical Circuit Switched (OCS) networks, an optimized Packet Loss Ratio (PLR), delay and Packet Delay Variation (PDV) performance can be obtained. It is, however, very demanding to offer a QoS comparable to OCS network through the OPS network in order to support a significant and growing number of applications. Thus, moving from OCS to OPS network, QoS is a critical issue. Hence, this paper analyzes the QoS requirements of network-intensive applications and services. Moreover, the paper derived an expression for performance metrics of a single OPS switch and ring OPS network. It is found that the International Telecommunication Union - Telecommunication (ITU-T) classification is extended with the new requirements of applications and services. For example, the upper bound delay, PDV and PLR of video streaming requires (100 ms, 400 ms), 50 ms and 1E−05 respectively. In this work, these values are improved as delay within 10 ms, PDV in micro seconds and PLR of 1E−07 at full load. It is also shown that a Fixed Delay Line (FDL) has to be considered when low PLR is required and when PDV and delay are not critical issues.

Performance Evaluation of Ethernet Network for Mobile Fronthual Networks

<p>Increasing mobile data traffic due to the rise of both smartphones and tablets has led to high-capacity demand of mobile data network. To meet the ever-growing capacity demand and reduce the cost of mobile network components, Cloud Radio Access Network (C-RAN) has emerged as a promising solution. In such network, the mobile operator’s Remote Radio Head (RRH) and Base Band Unit (BBU) are often separated and the connection between them has very tight timing and latency requirements imposed by Common Public Radio Interface (CPRI) and 3rd Generation Partnership Project (3GPP). This fronthaul connection is not yet provided by packet based network. To employ packet-based network for C-RAN fronthaul, the carried fronthaul traffic are needed to achieve the requirements of fronthaul streams. For this reason, the aim of this paper is focused on investigating and evaluating the feasibility of Ethernet networks for mobile fronthaul. The fronthaul requirements used to evaluate and investigate this network are maximum End to End (E2E) latency, Packet Loss Ratio (PLR) and Packet Delay Variation (PDV). The simulated results and numerical analysis confirm that the PDV and PLR of High Priority (HP) traffic in Ethernet network meet the requirements of mobile fronthaul using CPRI. However, the PDV of HP traffic meets the fronthaul network when the number of nodes in the Ethernet network is at most four. For Ethernet network, the number of nodes in the network limits the maximum separation distance between BBU and RRH (link length); for increasing the number of nodes, the link length decreases. Consequently, Radio over Ethernet (RoE) traffic should receive the priority and Quality of Service (QoS) HP can provide. On the other hand, Low Priority (LP) classes are not sensitive to QoS metrics and should be used for transporting time insensitive applications and services.</p>

Precision Packet-Based Frequency Transfer Based on Oversampling

Frequency synchronization of a distributed measurement system requires the transfer of an accurate frequency reference to all nodes. The use of a general-purpose packet-based network for this aim is analyzed in this paper, where oversampling is considered as a means to counter the effects of packet delay variation on time accuracy. A comprehensive analysis that includes the stability of the local clock is presented and shows that frequency transfer through a packet network of this kind is feasible, with an accuracy level that can be of interest to a number of distributed measurement applications.

Performance Analysis of Voice and Video Conference over DiffServ and MPLS

Multiprotocol Label Switching (MPLS) and Differentiated Services (DiffServ) play a vital role for minimizing the congestion by efficient load balancing and management of the network resources used to support Quality of Service (QoS). The lower network delay and efficient forwarding mechanism enhancing the speed of packet transfer make it more suitable for implementing real-time applications such as Voice and Video Conferencing. This paper evaluates the performance measures such as Packet Delay Variation (PDV) and end-to-end delay when applying some case studies comprising numbers of routers serving different number of clients. Both (PDV) and (End-to-End Delay) are determined using combination of Open Shortest Path First (OSPF), DiffServ and MPLS protocols. The results may lead to the measures of selection of the appropriate protocol.

Performance Analysis of Route Redistribution among Diverse Dynamic Routing Protocols based on OPNET Simulation

Routing protocols are the fundamental block of selecting the optimal path from a source node to a destination node in internetwork. Due to emerge the large networks in business aspect thus; they operate diverse routing protocols in their infrastructure. In order to keep a large network connected; the implementation of the route redistribution is needed in network routers. This paper creates the four scenarios on the same network topology by using Optimized Network Engineering Tools Modeler (OPNET 14.5) simulator in order to analyze the performance of the route redistribution among three routing protocols by configuring three protocols from a set of Routing Information Protocol (RIPv2), Enhanced Interior Gateway Routing Protocol (EIGRP), Open Shortest Path First (OSPF), and Intermediate System to Intermediate System (IS-IS) dynamic routing protocols on each scenario. The first scenario is EIGRP_OSPF_ISIS, the second scenario is EIGRP_OSPF_RIPv2, the third scenario is RIPv2_EIGRP_ISIS, and the fourth scenario is RIPv2_OSPF_ISIS. The simulation results showed that the RIPv2_EIGRP_ISIS scenario outperforms the other scenarios in terms of network convergence time, the hops number, jitter, packet delay variation, packet end to end delay; therefore, it fits real time applications such as voice and video conferencing. In contrast, the EIGRP_OSPF_ISIS scenario has better results compared with other scenarios in terms of response time in case of using web browsing, database query, and Email services.

Delay Minimization in Real-Time Communications With Joint Buffering and Coding

We present a closed-form expression for the minimal delay that is achievable in a setting that combines a buffer and an erasure code, used to mitigate the packet delay variance. The erasure code is modeled according to the recent information-theoretic results on finite block length codes. Evaluations reveal that accurate knowledge of the network parameters is essential for optimal operation. Moreover, it is shown that, when the network packet delay variance is large, the buffer delay becomes negligible. Therefore, in this case, the delay budget should be spent mainly on the erasure code.

A Case Study of IPv6 Network Performance: Packet Delay, Loss, and Reordering

Internet Protocol (IP) is used to identify and locate computers on the Internet. Currently, IPv4 still routes most Internet traffic. However, with the exhausting of IPv4 addresses, the transition to IPv6 is imminent, because, as the successor of IPv4, IPv6 can provide a larger available address space. Existing studies have addressed the notion that IPv6‐centric next generation networks are widely deployed and applied. In order to gain a deep understanding of IPv6, this paper revisits several critical IPv6 performance metrics. Our extensive measurement shows that packet delay and loss rate of IPv6 are similar to IPv4 when the AS‐level paths are roughly the same. Specifically, when the link utilization exceeds a threshold, for example, 0.83 in our study, variation of packet delay presents a similar pattern with the variation of link utilization. If packet delay of a path is large, packet‐loss rate of that path is more likely to fluctuate. In addition, we conduct a first‐ever analysis of packet reordering in IPv6 world. Few IPv6 probe packets are out‐of‐order and the reordering rate is , which is much lower than that of 0.79% in IPv4 world. Our analysis consolidates an experimental basis for operators and researchers of IPv6 networks.

Pktgen: Measuring performance on high speed networks

Pktgen is a tool for high-speed packet generation and testing. It runs in the Linux kernel, and is designed to accommodate a wide range of network performance tests. Pktgen consists of a packet generator, a receiver, and a protocol that defines the format for test packets. This paper describes the design of pktgen, and discusses its usage as a capable performance testing tool. The design is focused on performance, and in order to generate packets at high packet rate, pktgen takes advantage of multicore systems and multi-queue features on modern network cards. To support realistic heterogeneous traffic patterns, pktgen can generate multiple simultaneous UDP flows. Pktgen supports measurements of throughput, latency, packet inter-arrival time, and packet delay variation with a wide-range of fine-grained user specified rates.We evaluate and compare pktgen to other tools, and show that it is more than an order of magnitude faster than current Linux applications such as Iperf and Netperf, and has the same upper limit on performance as special-purpose high-speed tools, such as DPDK and Netmap, when it comes to throughput measurements with user-specified rate.

Calculation of packet jitter for non-poisson traffic

The packet delay variation, commonly called delay jitter, is an important quality of service parameter in IP networks especially for real-time applications. In this paper, we propose the exact and approximate models to compute the jitter for some non-Poisson FCFS queues with a single flow that are important for recent IP network. We show that the approximate models are sufficiently accurate for design purposes. We also show that these models can be computed sufficiently fast to be usable within some iterative procedure, e.g., for dimensioning a playback buffer or for flow assignment in a network.

An enhanced multiinterface multichannel algorithm for high quality live video streaming over hybrid WMNs

Recently, providing live video streaming over hybrid wireless mesh networks (WMNs) has been of great interest. In this kind of network, most of the users usually move with low speed and always keep their connections within the network by attaching to different mesh routers (MRs). When evaluating a mesh technology, it is important to pay special attention to this kind of device as it is mostly handled by thenal user. These devices, known as STA nodes, neither run a routing protocol nor perform forwarding operations. Although our proposed distributed locator service in GREENIE makes hybrid WMNs more efficient for supporting STA nodes, video distortion increases when the number of them sharply augments. In order to address this problem, we equip MRs with an efficient multiinterface multichannel (MIMC) algorithm. The proposed algorithm is an engineering solution that completely considers the channel status and its queue length. Moreover, it is cost effective and simple in its implementation without imposing unnecessary complexity on the network. Simulation results using OMNET++ show that all the tested routing protocols, and GREENIE in particular, efficiently exploit the advantages of the designed MIMC algorithm by improving the video distortion, total capacity of successfully received video packets, number of successfully received video frames, packet delay variation, and end-to-end delay metrics. As a result, better video quality can be provided on STA nodes.

Filtration model for the detection of malicious traffic in large-scale networks

This study proposes a capable, scalable, and reliable edge-to-edge model for filtering malicious traffic through real-time monitoring of the impact of user behavior on quality of service (QoS) regulations. The model investigates user traffic, including that injected through distributed gateways and that destined to gateways that are experiencing actual attacks. Misbehaving traffic filtration is triggered only when the network is congested, at which point burst gateways generate an explicit congestion notification (ECN) to misbehaving users. To investigate the behavior of misbehaving user traffic, packet delay variation (PDV) ratios are actively estimated and packet transfer rates are passively measured at a unit time. Users who exceed the PDV bit rates specified in their service level agreements (SLAs) are filtered as suspicious users. In addition, suspicious users who exceed the SLA bandwidth bit rates are filtered as network intruders. Simulation results demonstrate that the proposed model efficiently filters network traffic and precisely detects malicious traffic.

Adaptive Congestion Control for Unpredictable Cellular Networks

Legacy congestion controls including TCP and its variants are known to perform poorly over cellular networks due to highly variable capacities over short time scales, self-inflicted packet delays, and packet losses unrelated to congestion. To cope with these challenges, we present Verus, an end-to-end congestion control protocol that uses delay measurements to react quickly to the capacity changes in cellular networks without explicitly attempting to predict the cellular channel dynamics. The key idea of Verus is to continuously learn a delay profile that captures the relationship between end-to-end packet delay and outstanding window size over short epochs and uses this relationship to increment or decrement the window size based on the observed short-term packet delay variations. While the delay-based control is primarily for congestion avoidance, Verus uses standard TCP features including multiplicative decrease upon packet loss and slow start. Through a combination of simulations, empirical evaluations using cellular network traces, and real-world evaluations against standard TCP flavors and state of the art protocols like Sprout, we show that Verus outperforms these protocols in cellular channels. In comparison to TCP Cubic, Verus achieves an order of magnitude (> 10x) reduction in delay over 3G and LTE networks while achieving comparable throughput (sometimes marginally higher). In comparison to Sprout, Verus achieves up to 30% higher throughput in rapidly changing cellular networks.

Clock Skew Estimation Using Kalman Filter and IEEE 1588v2 PTP for Telecom Networks

Accurate and precise frequency synchronization is an essential requirement for different areas in telecommunication industry. Emerging practise for frequency synchronization is to utilize packet networks since it is highly cost effective. One method of distributing frequency over a variety of packet networks is based on well known IEEE 1588v2 PTP standard that uses a master-slave architecture. Accuracy and precision of the frequency synchronization over IEEE 1588v2 PTP is mainly deteriorated by the Packet Delay Variations (PDVs) experienced on the transmission path. This problem can be overcome by deploying so-called timing aware routing elements, however slaves' frequency accuracy and precision still heavily relies on the quality of the slave's clock synchronization algorithm. Hence, this work introduces an improved Kalman filter based algorithm for frequency synchronization over IEEE 1588v2 protocol that outperforms other prior art techniques. The algorithm's performance is evaluated using simulation, yet the network impairments (PDVs) are based on experimentally measured data.

A new metric for flow‐level filtering of low‐rate DDoS attacks

AbstractLow‐rate distributed denial‐of‐service (LDDoS) attacks dramatically reduce transmission control protocol throughput by exploiting the vulnerability in the transmission control protocol congestion control mechanism. The current study proposes a new metric called mean Internet Protocol (IP) packet delay variation (mipdv) to detect LDDoS flows and a filtering method called ipdv‐based LDDoS filtering (ILF) using mipdv. Receiving first seven packets from a flow is sufficient to calculate the mipdv metric. Subsequently, mipdv can be recalculated for each received packet. This makes the detection of LDDoS flows possible in a short time (in a few tens of milliseconds in most cases). Ns2 simulations were conducted to evaluate the performance of ILF. Experimental results show that ILF detects LDDoS flows in a very short time with very high accuracy. Copyright © 2015 John Wiley & Sons, Ltd.

Fisher Information-based Experiment Design for Network Tomography

Network tomography aims to infer the individual performance of networked elements (e.g., links) using aggregate measurements on end-to-end paths. Previous work on network tomography focuses primarily on developing estimators using the given measurements, while the design of measurements is often neglected. We fill this gap by proposing a framework to design probing experiments with focus on probe allocation, and applying it to two concrete problems: packet loss tomography and packet delay variation (PDV) tomography. Based on the Fisher Information Matrix (FIM), we design the distribution of probes across paths to maximize the best accuracy of unbiased estimators, asymptotically achievable by the maximum likelihood estimator. We consider two widely-adopted objective functions: determinant of the inverse FIM (D-optimality) and trace of the inverse FIM (A-optimality). We also extend the A-optimal criterion to incorporate heterogeneity in link weights. Under certain conditions on the FIM, satisfied by both loss and PDV tomography, we derive explicit expressions for both objective functions. When the number of probing paths equals the number of links, these lead to closed-form solutions for the optimal design; when there are more paths, we develop a heuristic to select a subset of paths and optimally allocate probes within the subset. Observing the dependency of the optimal design on unknown parameters, we further propose an algorithm that iteratively updates the design based on parameter estimates, which converges to the design based on true parameters as the number of probes increases. Using packet-level simulations on real datasets, we verify that the proposed design effectively reduces estimation error compared with the common approach of uniformly distributing probes.