Analytical Analysis of Applying Packet Fragmentation Mechanism on Both Basic and RTS/CTS Access Methods of the IEEE 802.11b DCF Network Under Imperfect Channel and Finite Load Conditions

The mathematical modeling and performance evaluation of the IEEE 802.11 network in all its various extensions (802.11b, 802.11a, 802.11g, 802.11e, 802.11n, etc.) have already been widely explored over the past years. However, the Packet Fragmentation Mechanism (PFM), which is proposed by the IEEE work group to enhance the MAC sub-layer of the IEEE 802.11 standard in an error-prone channel, has been missed in the available literature. Yet, the PFM is the only existing solution to reduce the influence of bit error rate and the length of data packets on the packet error rate, and consequently on the performances of IEEE 802.11 networks. In this paper, we propose a new three-dimensional Markov chain in order to model, for the first time in the literature, the PFM in both Basic and RTS/CTS access methods of the IEEE 802.11b DCF network under imperfect channel and finite load conditions. Then, we develop mathematical models to derive a variety of performance metrics, such as: the overall throughput, the average packet delay successfully transmitted, the average packet drop time, the delay jitter and the packet delay distribution. Performance analysis of applying PFM on both Basic and RTS/CTS access methods of the IEEE 802.11b DCF network under imperfect channel and finite load conditions shows original results and leads to new conclusions that could not be intuitively expected.     

A Pseudo-Bayesian ALOHA Algorithm with Mixed Priorities

In reservation-based multiple access protocols, before obtaining a contention-free access to the channel, a mobile terminal must wait for its request packet to be successfully sent to the base station. A pseudo-Bayesian ALOHA algorithm with multiple priorities is proposed in this paper to reduce the waiting time of delay sensitive request packets in a multimedia environment. Packets are transmitted in each slot according to a transmission probability based on the channel history and a priority parameter assigned to their respective priority class. An adaptation of the slotted protocol to the framed environment proposed for wireless ATM is also described. Simulation results are presented to show that the protocol offers a significant delay improvement for high priority packets with both Poisson and self-similar traffic while low priority packets only experience a slight performance degradation.     

一种基于游标的多径流量分割算法

 多径传输使用多条连接源节点和目的节点的路径进行传输,在提高资源利用率的同时会引起包乱序问题,并且存在路径之间的负载均衡问题.本文提出了一种基于游标的流量分割算法,游标是当前路径传输延迟与相邻包到达源节点的时间差之间的差值,它作为选取路径的延时基线来保证包到达的有序性,游标会随着路径不同或相邻包到达源节点的时间差不同而动态地滑动,通过动态滑动游标使得尽可能多的路径可用来传输当前包,从而很好地实现负载均衡.仿真结果表明,与已有的保证包有序的算法相比,本算法使负载更加均衡.     

Measurement and analysis of single-hop delay on an IP backbone network

We measure and analyze the single-hop packet delay through operational routers in the Sprint Internet protocol (IP) backbone network. After presenting our delay measurements through a single router for OC-3 and OC-12 link speeds, we propose a methodology to identify the factors contributing to single-hop delay. In addition to packet processing, transmission, and queueing delay at the output link, we observe the presence of very large delays that cannot be explained within the context of a first-in first-out output queue model. We isolate and analyze these outliers. Results indicate that there is very little queueing taking place in Sprint's backbone. As link speeds increase, transmission delay decreases and the dominant part of single-hop delay is packet processing time. We show that if a packet is received and transmitted on the same linecard, it experiences less than 20 /spl mu/s of delay. If the packet is transmitted across the switch fabric, its delay doubles in magnitude. We observe that processing due to IP options results in single-hop delays in the order of milliseconds. Milliseconds of delay may also be experienced by packets that do not carry IP options. We attribute those delays to router idiosyncratic behavior that affects less than 1% of the packets. Finally, we show that the queueing delay distribution is long-tailed and can be approximated with a Weibull distribution with the scale parameter a=0.5 and the shape parameter b=0.6 to 0.82.     

基于路径时延模型的车联网数据分发方案

针对车联网的容迟特性造成通信资源受限的问题,提出了满足副本抑制要求的数据分发方案.方案利用马尔可夫链,通过交通网络的车辆概率分布建立路段的期望传输时延,并结合车辆的轨迹与目标位置的匹配度确定车辆的转发优先级.车辆为转发的每个数据包插入转发参数字段并通过同步反馈机制确定最终的转发车辆,确保由优先级最高的车辆完成转发.考虑到链路的稳定性,还推导了当前丢包率前提下,车辆接收数据包与发送次数之比,避免不必要的发送尝试产生大量副本.实验结果显示,提出的方案与基于轨迹预测的算法相比,有效提高了网络吞吐量和时延性能.     

Packet Delay Statistics of the Multichannel Selective-Repeat Automatic-Repeat-Request

In this paper, we conduct stochastic modeling and analysis of the packet end-to-end delay in a multichannel selective-repeat automatic-repeat-request (MSR-ARQ) protocol. In this protocol, the transmitter continuously transmits packets over multiple parallel channels and retransmits erroneously received packets with either dynamic or static packet-to-channel scheduling policy. Under the assumption that packets are always supplied at the transmitter, denoted by the saturated traffic condition, we analyze the steady state probability distribution function of the delay of an arbitrary packet, which is measured by the duration between the instant at which the packet is transmitted for the first time and the time it departs from the resequencing queue at the receiver. Using the analysis result, we numerically compute the distribution function for chosen values of the number of channels and the error rates to demonstrate the computational effectiveness of the result. With numerical and simulation results, we then study the performance of MSR-ARQ in terms of the mean packet delay and compare the two scheduling policies. It is shown that the dynamic scheduling achieves a better packet delay performance than the static scheduling. With the dynamic scheduling and the presence of difference between the error rates of parallel channels, the mean packet delay decreases as the difference between channels?? error rates increases. Moreover, the number of parallel channels has an insignificant impact on the mean packet delay, which shows that the use of parallel channels is favorable for the wireless or mobile communications to increase the data transmission rate while keeping the mean packet delay at an acceptable level.     

Performance evaluation of CSMA/ID MAC protocol for IP over WDM ring networks

In this paper, a packet pre‐classification media access control protocol based on a carrier sense multiple access with idle detection (CSMA/ID) scheme is investigated for supporting IP packets over all‐optical WDM ring networks. The purpose of the protocol is to increase throughput and to decrease the packet transmission delay of IP packets over optical networks in a metropolitan area network. This protocol avoids both packet collision and packet fragmentation. In order to improve the utilization of the network, the packets transmitted from a local area network are first pre‐classified into various class queues of an access point (AP) according to their length. After checking the available space based on the wavelength received by the receivers of the AP, the packets in the queues are transmitted. An analytical model is developed to evaluate the performance of the protocol, with simulation results showing good network efficiency. The proposed network has short‐term variations that introduce unfairness conditions. This problem could be overcome by assigning a quota on individual queues to allow all queues fair access. Copyright © 2008 John Wiley & Sons, Ltd.     

Resource allocation for multiple classes of DS-CDMA traffic

We consider a packet data direct-sequence code-division multiple-access (DS-CDMA) system which supports integrated services. The services are partitioned into different traffic classes according to information rate (bandwidth) and quality of service (QoS) requirements. Given sufficient bandwidth, QoS requirements can be satisfied by an appropriate assignment of transmitted power and processing gain to users in each class. The effect of this assignment is analyzed for both a single class of data users and two classes of voice and data users. For a single class of data users, we examine the relationship between average delay and processing gain, assuming that ARQ with forward error correction is used to guarantee reliability. The only channel impairment considered is interference, which is modeled as Gaussian noise. A fixed user population is assumed and two models for generation of data packets are considered: (1) each user generates a new packet as soon as the preceding packet is successfully delivered and (2) each user generates packets according to a Poisson process. In each case, the packets enter a buffer which is emptied at the symbol rate. For the second traffic model, lowering the processing gain below a threshold can produce multiple operating points, one of which corresponds to infinite delay. The choice of processing gain which minimizes average delay in that case is the smallest processing gain at which multiple operating points are avoided. Two classes of users (voice/data and two data classes) are then considered. Numerical examples are presented which illustrate, the increase in the two-dimensional (2-D) capacity region achievable by optimizing the assignment of powers and processing gains to each class     

Performance Analysis of a Quasi-M/D/1 Cut-Through Switching Network with Noisy Channels

Cut-through switching is advantageous in that it can reduce the transmission delay compared with the conventional message or packet switching. In this paper, when the channel is noisy, we investigate various properties ofM/D/1quasi-cut-through switching including actual traffic intensity and overall network delay. In the analysis of delay resulting from retransmission of erroneous packets, we have included the average transmission time of negative acknowledgment signal and queueing time for the retransmitted packet so that the overall network delay can be obtained accurately. In addition, we have obtained distributions of the number of nodes to be traversed and the number of nodes through which packets pass by cut. According to the analysis results, the performance of cut-through switching is superior to that of conventional packet switching in most practical ranges of parameter values.     

Energy considerations for topology-unaware TDMA MAC protocols

Since the energy budget of mobile nodes is limited, the performance of a networking protocol for such users should be evaluated in terms of its energy efficiency, in addition to the more traditional metrics such as throughput. In this paper, two topology-unaware MAC protocols—in which the scheduling time slots are allocated irrespectively of the underline topology—are considered and their energy consumption is derived. It turns out that the per frame power consumption is lower for the less throughput-efficient protocol, suggesting that energy savings are achieved at the expense of throughput.A finer energy consumption study is carried out in the sequel, focusing on the amount of energy consumed to successfully transmit a certain number of packets, or equivalently, on the per successful transmission power consumption. It is shown that the more throughput-efficient protocol consumes less energy per successful transmission under certain conditions (which are derived), due to the lower number of transmission attempts before a data packet is successfully transmitted. The same energy-efficiency relation is observed under certain conditions (which are derived) when data packets are delay constrained and, thus, may become obsolete if not transmitted successfully within a specific time interval. The conditions under which the per successful transmission power consumption is minimized for delay-constrained packets, are also established in this work and it is observed that when the system throughput is maximized, the power consumed is close to the minimum. Simulation results support the claims and the expectations of the aforementioned analysis.     

Low latency handover scheme for 5G dual-connectivity scenario

A novel handover scheme was proposed for the secondary node (SN) unchanged 5G dual-connectivity scenarios.In the proposed novel scheme,the SN connection was maintained for data packet transmission during the handover,however,both the main node (MN) and the SN were completely disconnected in the legacy scheme.The transmission delay during handover was decreased greatly by the proposed scheme.Firstly,the legacy handover scheme was analyzed and its deficiency was figured out.Then,the novel mechanism’s signaling interaction was elaborated and the time sequence models for the novel scheme and the legacy scheme were further established.Finally,based on the time sequence model,the performance evaluation processes were carried out in terms of mathematical modeling and experimental simulations.The analysis results demonstrate that the proposed novel scheme reduces the single packet transmission delay,the average transmission delay and the total transmission delay,and has good performance advantages.     

Analysis and heuristics for the characterization of selective repeat ARQ delay statistics over wireless channels

We consider a point-to-point wireless transmission where link layer ARQ is used to counteract channel impairments. In particular, we refer, as an example, to a 3G cellular system, where a dedicated channel is used between a mobile terminal and its serving base station. Our aim is to find accurate and fast heuristics for the characterization of link layer and higher level (e.g., application level) packet delay. Existing methods to obtain such statistics are often based on recursive computations or large-sized matrix manipulations. For these reasons, they are too complex to be successfully applied in a mobile terminal due to memory, delay, and energy constraints. We first present an analytical framework to compute link-layer packet delivery delay statistics as a function of the packet error rate; then we extend the model in order to find the statistics related to higher level packets (i.e., to aggregates of link layer packets). Both in-order and out-of-order delivery of link-layer packets to higher levels are considered. The goodness of the channel model considered in the analysis is proved by means of accurate channel simulation results. The obtained statistics are then characterized by highlighting their properties as a function of the round-trip time and the error rate at the link layer. Finally, fast and accurate heuristics are derived directly from the analysis. These heuristics are very simple (piecewise linear functions), so they can be effectively used in a mobile terminal to obtain accurate delay statistics estimates with little computational effort.     

Scheduling nonuniform traffic in a packet switching system withlarge propagation delay

Transmission algorithms are introduced for use in a single-hop packet switching system with nonuniform traffic and with propagation delay that Is large relative to the packet transmission time. The traffic model allows arbitrary traffic streams subject only to a constraint on the number of data packets which can arrive at any individual source in the system or for any individual destination in the system over time periods of specified length. The algorithms are based primarily on sending transmission schedules to the receivers immediately before transmitting each data packet multiple times so that the receiver can maximize the number of packets it captures. An algorithm based on matchings in a random graph is shown to provide mean total delay divided by mean propagation delay arbitrarily close to one, as the propagation delay tends to infinity     

Power allocation and transmission scheduling for a network with bidirectional relaying links

In this paper, we study packet transmission scheduling for a network with bidirectional relaying links, where the relay station can use network coding to combine packets to multiple receivers and opportunistically decide the number of packets to be combined in each transmission. Two cases are considered, depending on whether nodes are allowed to overhear transmissions of each other. A constrained Markov decision process is first formulated with an objective to minimize the average delay of packet transmissions, subject to the maximum and average transmission power limits of the relay node. The complexity for solving the constrained Markov decision process (MDP) is prohibitively high, although the computational complexity for the no‐overhearing case can be greatly reduced. Heuristic schemes are then proposed, one applies to the general case, and another applies to only the no‐overhearing case. Numerical results demonstrate that the heuristic schemes can achieve close‐to‐optimum average packet transmission delay, and furthermore, the second scheme achieves lower maximum delay while keeping the same average packet transmission delay and relay node power consumption as the first one. Copyright © 2015 John Wiley & Sons, Ltd.     

Frequency-Domain Stochastic Error Concealment for Wireless Audio Applications

Real time digital audio delivery over Wireless Local Area Networks (WLANs) represents an attractive, flexible and cost effective framework for realizing high-quality, multichannel home audio applications. However, the unreliable nature of WLANs IP link frequently imposes significant playback quality degradation, due to delay or permanent loss of a number of transmitted digital audio packets. In this paper, a novel packet error concealment technique is presented, based on the spectral reconstruction of the statistical equivalent of a previously successfully received audio data packet. It is shown that the proposed data reconstruction scheme outperforms previously published error concealment strategies, in both terms of objective and perceptual criteria.     

A channel reuse strategy with adaptive channel allocation for all-optical WDM networks

The fundamental problems of WDM networks are: (1) high rate of control packet loss and (2) high propagation delay for each (re)transmission. In this paper, we minimize the station randomness to access the control architecture introducing a collisions-free access scheme. We propose a synchronous protocol according which at the end of the propagation delay each station applies a distributed algorithm for packet transmission following the data channel collisions and the receiver collisions avoidance algorithms. We introduce two data transmission stages. The time difference between them is one packet transmission time. At the end of the first stage all data channels are free and can be reused by the remaining data packets during the second stage. The proposed protocol ensures a totally collisions-free performance. The main advantage is that the data channels reuse strategy applied during the second stage provides enhanced transmission probability to the rejected packets during the first stage. This allows the data packets to try retransmission in the same cycle without requiring control packets re-coordination that increases propagation delay. Thus, we achieve large number of data packets transmission, even more than the data channels number, providing throughput improvement and delay reduction, comparing with other studies.     

Interleaved coding and sequential transmission scheme for packet‐level forward error correction over burst loss channels

Burst packet loss is a common problem over wired and wireless networks and leads to a significant reduction in the performance of packet‐level forward error correction (FEC) schemes used to recover packet losses during transmission. Traditional FEC interleaving methods adopt the sequential coding‐interleaved transmission (SCIT) process to encode the FEC packets sequentially and reorder the packet transmission sequence. Consequently, the burst loss effect can be mitigated at the expense of an increased end‐to‐end delay. Alternatively, the reversed interleaving scheme, namely, interleaved coding‐sequential transmission (ICST), performs FEC coding in an interleaved manner and transmits the packets sequentially based on their generation order in the application. In this study, the analytical FEC model is constructed to evaluate the performance of the SCIT and ICST schemes. From the analysis results, it can be observed that the interleaving delay of ICST FEC is reduced by transmitting the source packets immediately as they arrive from the application. Accordingly, an Enhanced ICST scheme is further proposed to trade the saved interleaving time for a greater interleaving capacity, and the corresponding packet loss rate can be minimized under a given delay constraint. The simulation results show that the Enhanced ICST scheme achieves a lower packet loss rate and a higher peak signal‐to‐noise‐ratio than the traditional SCIT and ICST schemes for video streaming applications.     

一种适用于虚拟机网络的数据包高效传输方法

为了提高数据包在云计算数据中心中基于虚拟机构成网络中的传输性能,提出了一种基于网络编码的高效数据包传输方法.基于网络编码机制,采用对传输过程中丢失数据包高效的编码组合策略,多个虚拟机终端可以在一次多播或广播传输中获取多个从交换机优先传输的数据包,因此,提出的方法可以提高基于虚拟机网络的多播及广播业务的数据包传输延迟,并提高多播及广播业务的网络吞吐量.仿真结果表明提出的方法在典型信道条件下均获得了较好的数据包传输时延及网络吞吐量性能.     

Burst transmission algorithm to improve packet level performance in contention-free slotted OBS networks

In contention-free slotted optical burst switching (SOBS) networks, controllers are utilized in order to manage the time-slot assignment, avoiding congestions among multiple burst transmissions. In this network, bursts are never lost at intermediate nodes but packets are lost at an ingress edge node due to a burst transmission algorithm. In addition, packet transmission delay increases depending on the algorithm. In order to improve packet level performance, in this paper, we propose a new burst transmission algorithm. In this method, two different thresholds are used; one is used to send a control packet to a controller and the other is used to assemble a burst. With these thresholds, a time slot can be assigned to a burst in advance and packet level performance can be improved. In order to evaluate its packet level performance and investigate the impact of thresholds, we also propose a queueing model of a finite buffer where a batch of packets are served in a slot of a constant length. Numerical results show that our proposed method can decrease packet loss probability and transmission delay with two thresholds. In addition, we show that our analysis results are effective to investigate the performance of the proposed method when the number of wavelengths is large.     

An optical router with multistage management functions for asynchronous optical packet switching network

An optical router with multistage distributed management features for the asynchronous optical packet switching (OPS) network is presented, which can improve switching capacity and all-optical scalability. A compact recycling-fiber-delay-line (Rec-FDL) based collision resolution mechanism is proposed to resolve the contentions for asynchronous and variable length optical packets. The analysis models of stabilities, packet loss rates (PLR) and average packet waiting latencies (PWL) for the router are developed based on the timer based optical packet assembly algorithm. The simulation shows that PLR and PWL for a 400-byte optical packet transmitted in the 32 wavelengths dense wavelength division multiplexing (DWDM) system equal to 3.48 × 10⁻⁴ and 0.072 ns, respectively. The non-blocking switching can be realized for the packets with lengths less than the buffer granularity of the Rec-FDL, and the optimized performance for the proposed router can be obtained through properly selecting of the system parameters.