Packet level acknowledgement and Go‐Back‐N protocol performance in infrared wireless LANs

Infrared wireless LANs may employ repetition rate (RR) coding to increase the symbol capture probability at the receiver. This paper examines the effectiveness of RR coding to utilization for infrared LANs using the physical and link layer parameter values proposed in the Advanced Infrared (AIr) protocol standard, which is developed by the Infrared Data Association (IrDA). Infrared LANs employ a Go‐Back‐N (GBN) automatic repeat request (ARQ) retransmission scheme at the Link Control (LC) layer to ensure reliable information transfer. To efficiently implement RR coding, the receiver may return after every DATA packet a suggestion for the suitable RR value to be used by the transmitter and implement a Stop‐and‐Wait (SW) ARQ scheme at the medium access control (MAC) layer. The effectiveness of employing this optional SW ARQ scheme at the MAC layer is discussed. Analytical models for the ARQ retransmission schemes are developed and employed to compare protocol utilization for different link parameter values such as window size, packet length and LC time out periods. This analysis identifies the ARQ protocol that maximizes performance for the specific link quality and the implemented link layer parameters. The effectiveness of the proposed RR coding to LAN utilization for different ARQ scheme implementation is finally explored. This analysis identifies the link quality level at which RR should be adjusted for maximum performance. It is concluded that if the packet error rate is higher than 0.1–0.4 (depending on the implemented ARQ protocol), the receiver should advise the transmitter to double the implemented RR for maximum performance. These error rate values are high and can be effectively estimated by the transmitter based on packet retransmissions. Thus, the usefulness of the receiver indicating to the transmitter to adjust RR is questionable, as the transmitter can effectively implement the suitable RR value based on packet retransmissions. Copyright © 2003 John Wiley & Sons, Ltd.     

无线通信中流媒体延迟的研究

尽管移动商务的迅速发展对无线网络提出了传输流媒体的新需求,但无线网络由于信道质量不稳定,传输错误率较高,这给流媒体传输服务质量(QoS)的保证提出了极大的挑战。对无线通信网络中流媒体传输延迟的最小化进行了讨论,并基于自动请求重传(ARQ)和前向差错控制(FEC)方法,推导出无线网络最小延迟时的最佳发送包的大小和误码率的关系,提出了延迟最小化的最佳拆分包长和最小带宽的算法,并通过实例得出误码率BER=10-3对延迟是个重要的界限。     

Providing deterministic packet delays and packet losses in multimedia wireless networks

‘Anytime, anywhere’ communication, information access and processing are much cherished in modern societies because of their ability to bring flexibility, freedom and increased efficiency to individuals and organizations. Wireless communications, by providing ubiquitous and tetherless network connectivity to mobile users, are therefore bound to play a major role in the advancement of our society. Although initial proposals and implementations of wireless communications are generally focused on near‐term voice and electronic messaging applications, it is recognized that future wireless communications will have to evolve towards supporting a wider range of applications, including voice, video, data, images and connections to wired networks. This implies that future wireless networks must provide quality‐of‐service (QoS) guarantees to various multimedia applications in a wireless environment. Typical traffic in multimedia applications can be classified as either Constant‐Bit‐Rate (CBR) traffic or Variable‐Bit‐Rate (VBR) traffic. In particular, scheduling the transmission of VBR multimedia traffic streams in a wireless environment is very challenging and is still an open problem. In general, there are two ways to guarantee the QoS of VBR multimedia streams, either deterministically or statistically. In particular, most connection admission control (CAC) algorithms and medium access control (MAC) protocols that have been proposed for multimedia wireless networks only provide statistical, or soft, QoS guarantees. In this paper, we consider deterministic QoS guarantees in multimedia wireless networks. We propose a method for constructing a packet‐dropping mechanism that is based on a mathematical framework that determines how many packets can be dropped while the required QoS can still be preserved. This is achieved by employing: (1) An accurate traffic characterization of the VBR multimedia traffic streams; (2) A traffic regulator that can provide bounded packet loss and (3) A traffic scheduler that can provide bounded packet delay. The combination of traffic characterization, regulation and scheduling can provide bounded loss and delay deterministically. This is a distinction from traditional deterministic QoS schemes in which a 0% packet loss are always assumed with deterministically bounding the delay. We performed a set of performance evaluation experiments. The results will demonstrate that our proposed QoS guarantee schemes can significantly support more connections than a system, which does not allow any loss, at the same required QoS. Moreover, from our evaluation experiments, we found that the proposed algorithms are able to out‐perform scheduling algorithms adopted in state‐of‐the‐art wireless MAC protocols, for example Mobile Access Scheme Based on Contention and Reservation for ATM (MASCARA) when the worst‐case traffic is being considered. Copyright © 2002 John Wiley & Sons, Ltd.     

Optimization of Hybrid ARQ for IP Packet Transmission

This paper presents, optimizes, and analyzes the performance of a novel hybridSelective Repeat/Multi Copy(SR/MC) Automatic Repeat Request (ARQ) scheme for transmitting fragmentedInternetProtocol (IP) packets. The ARQ scheme works in the SR mode until the last IPpacket fragment istransmitted. If a fragment is negatively acknowledged after the last fragmentis transmitted, then the system goes into theMC mode. In the MC mode, multiple copies of the erroneous fragment aretransmitted. After the IPfragments are received without error, the system returns to the SR mode.The optimization of the ARQ is done in terms of two parameters: fragment sizeand the optimum number of packetsto be transmitted in the MC mode, M. Optimum values for both parameters arecalculated for Bit ErrorRate (BER), throughput, IP packet size, and delay. The fragment size is alsocalculated for actual datathroughput for a given IP packet size, both with and without Forward ErrorCorrection (FEC). Then,the performance of the proposed scheme is evaluated in terms of BER andIP packet size with theoptimum M and fragment size. Performance results are obtained with and withoutBose ChaudhuriHocquenghem (BCH) error correction codes under Additive White Gaussian Noise(AWGN) as wellas Flat Rayleigh Fading channels. The ARQ scheme gives optimum performance forM equal to 10fragments and fragment size of 75 bytes. Under the AWGN channel, a throughputof 0.9 is achieved for any IPpacket size and at higher BER conditions compared to the Selective Repeat +Stutter Scheme 2 (SR + ST 2).An 8 dB improvement is achieved under the flat Rayleigh fading channel usingBCH(63, 51, 2) for a throughputof 0.9.     

End‐to‐end delay and packet drop rate performance for a wireless sensor network with a cluster‐tree topology

In this paper, we study the delay performance in a wireless sensor network (WSN) with a cluster‐tree topology. The end‐to‐end delay in such a network can be strongly dependent on the relative location between the sensors and the sink and the resource allocations of the cluster heads (CHs). For real‐time traffic, packets transmitted with excessive delay are dropped. Given the timeline allocations of each CH for local and inter‐cluster traffic transmissions, an analytical model is developed to find the distribution of the end‐to‐end transmission delay for packets originated from different clusters. Based on this result, the packet drop rate is derived. A heuristic scheme is then proposed to jointly find the timeline allocations of all the CHs in a WSN in order to achieve the minimum and balanced packet drop rate for traffic originated from different levels of the cluster tree. Simulation results are shown to verify the analysis and to demonstrate the effectiveness of the proposed CH timeline allocation scheme. Copyright © 2012 John Wiley & Sons, Ltd.     

Performance of H.263 video transmission over wireless channelsusing hybrid ARQ

This paper proposes a hybrid ARQ error control scheme based on the concatenation of a Reed-Solomon (RS) code and a rate compatible punctured convolutional (RCPC) code for low-bit-rate video transmission over wireless channels. The concatenated hybrid ARQ scheme we propose combines the advantages of both type-I and type-II hybrid ARQ schemes. Certain error correction capability is provided in each (re)transmitted packet, and the information can be recovered from each transmission or retransmission alone if the errors are within the error correction capability (similar to type-I hybrid ARQ). The retransmitted packet contains redundancy bits which, when combined with the previous transmission, result in a more powerful RS/convolutional concatenated code to recover information if error correction fails for the individual transmissions (similar to type-II hybrid ARQ). Bit-error rate (BER) or signal-to-noise ratio (SNR) of a radio channel changes over time due to mobile movement and fading. The channel quality at any instant depends on the previous channel conditions. For the accurate analysis of the performance of the hybrid ARQ scheme, we use a multistate Markov chain (MSMC) to model the radio channel at the data packet level. We propose a method to partition the range of the received SNR into a set of states for constructing the model so that the difference between the error rate of the real radio channel and that of the MSMC model is minimized. Based on the model, we analyze the performance of the concatenated hybrid ARQ scheme. The results give valuable insight into the effects of the error protection capability in each packet, the mobile speed, and the number of retransmissions. Finally, the transmission of H.263 coded video over a wireless channel with error protection provided by the concatenated hybrid ARQ scheme is studied by means of simulations     

Wireless Packet Scheduling Algorithm for OFDMA System Based on Time‐Utility and Channel State

In this paper, we propose an urgency‐ and efficiencybased wireless packet scheduling (UEPS) algorithm that is able to schedule real‐time (RT) and non‐real‐time (NRT) traffics at the same time while supporting multiple users simultaneously at any given scheduling time instant. The UEPS algorithm is designed to support wireless downlink packet scheduling in an orthogonal frequency division multiple access (OFDMA) system, which is a strong candidate as a wireless access method for the next generation of wireless communications. The UEPS algorithm uses the time‐utility function as a scheduling urgency factor and the relative status of the current channel to the average channel status as an efficiency indicator of radio resource usage. The design goal of the UEPS algorithm is to maximize throughput of NRT traffics while satisfying quality‐of‐service (QoS) requirements of RT traffics. The simulation study shows that the UEPS algorithm is able to give better throughput performance than existing wireless packet scheduling algorithms such as proportional fair (PF) and modifiedlargest weighted delay first (M‐LWDF), while satisfying the QoS requirements of RT traffics such as average delay and packet loss rate under various traffic loads.     

Effective Admission Policy for Multimedia Traffic Connections over Satellite DVB‐RCS Network

Thanks to the great possibilities of providing different types of telecommunication traffic to a large geographical area, satellite networks are expected to be an essential component of the next‐generation internet. As a result, issues concerning the designing and testing of efficient connection‐admission‐control (CAC) strategies in order to increase the quality of service (QoS) for multimedia traffic sources, are attractive and at the cutting edge of research. This paper investigates the potential strengths of a generic digital‐video‐broadcasting return‐channel‐via‐satellite (DVB‐RCS) system architecture, proposing a new CAC algorithm with the aim of efficiently managing real‐time multimedia video sources, both with constant and high variable data rate transmission; moreover, the proposed admission strategy is compared with a well‐known iterative CAC mainly designed for the managing of real‐time bursty traffic sources in order to demonstrate that the new algorithm is also well suited for those traffic sources. Performance analysis shows that, both algorithms guarantee the agreed QoS to real‐time bursty connections that are more sensitive to delay jitter; however, our proposed algorithm can also manage interactive real‐time multimedia traffic sources in high load and mixed traffic conditions.     

Reliability-based incremental redundancy with convolutional codes

Incremental redundancy, or Hybrid type-II ARQ (HARQ), algorithms use a combination of forward error correction and retransmissions to guarantee reliable packet data communications. In this work, we propose a HARQ algorithm that exploits received packet reliability to improve system performance. Specifically, the receiver uses the average magnitude of the log-likelihood ratios of the information bits as the packet reliability metric, which is then used to determine the sizes of subsequent retransmissions. The proposed retransmission strategy attempts to maximize user throughput while satisfying a maximum packet delay constraint. The performance of our reliability-based HARQ algorithm is evaluated in static and time-varying channels through simulations. Furthermore, analytical results on the relationship between the reliability metric, the code rate and the block error rate are presented.     

适用于卫星网络的TCP跨层改进机制

该文提出基于跨层信息交互,将链路层ARQ重传状态信息通知TCP的机制,避免了链路层重传引起的时延变化对TCP的不利影响。该机制使用完全可靠选择性重传ARQ为TCP提供可靠的链路,避免卫星链路上发生丢包,并且不必要求链路层保证包按序递交,消除了重排序的等待时延,适合带宽时延积较大的卫星网络。仿真结果表明,能显著提高TCP在卫星网中的性能,特别是在误帧率较高条件下。     

Performance Modeling of the Fast and Reliable Two-Layer Retransmission for CDMA Systems

1　Introduction　With the proliferation of the World Wide Web (WWW)in our daily life, a number of wireless data services[1] suchas voice, audio, video streaming, file and web downloadingalso need to be supported in the wireless access networks.To bring the WWW traffic to the wireless mobile devices, itis important that a suitable protocol or standard is chosen tocater to the growing demands of data services over wirelesschannels which could handle a wide variety of multimediatraffic with …     

An Energy Saving Scheme for Multilane‐Based High‐Speed Ethernet

In this paper, we propose a scheme for partially dynamic lane control for energy saving in multilane‐based high‐speed Ethernet. In this scheme, among the given transmission lanes, at least one lane is always operating, and the remaining lanes are dynamically activated to alleviate the network performance in terms of queuing delay and packet loss in the range of acceptance. The number of active lanes is determined by the decision algorithm based on the information regarding traffic and queue status. The reconciliation sublayer adjusts the transmission lane with the updated number of lanes received from the algorithm, which guarantees no processing delay in the media access control layer, no overhead, and minimal delay of the exchanging control frames. The proposed scheme is simulated in terms of queuing delay, packet loss rate, lane changes, and energy saving using an OPNET simulator. Our results indicate that energy savings of around 55% (or, when the offered load is less than 0.25, a significant additional savings of up to 75%) can be obtained with a queuing delay of less than 1 ms, a packet loss of less than 10^?4, and a control packet exchange time of less than 0.5 μs in random traffic.     

Scalable rate control for video transmission over UMTS

This paper proposes a mechanism for the congestion control for video transmission over Universal Mobile Telecommunications System (UMTS). Our scheme is applied when the mobile user experiences real time multimedia content and adopts the theory of a widely accepted rate control method in wired networks, namely equation‐based rate control. In this approach, the transmission rate of the multimedia data is determined as a function of the packet loss rate, the round trip time and the packet size and the server explicitly adjusts its sending rate as a function of these parameters. Through a number of simulations and experiments we validate the correctness and measure the performance and efficiency of the mechanism. A first level of evaluation is carried out using the ns‐2 simulator. A second level then validates the proposed mechanism in real world traffic scenarios by performing experiments in a commercial UMTS network. Copyright © 2006 John Wiley & Sons, Ltd.     

Performance analysis of adaptive MAC protocol in VANETs considering the potential impact on throughput and transmission delays

The existing adaptive multichannel medium access control (MAC) protocols in vehicular ad hoc networks can adjust themselves according to different vehicular traffic densities. These protocols can increase throughput and guarantee a bounded transmission delay for real‐time safety applications. However, the optimized control channel interval is computed based on the maximum throughput while ignoring the strict safety packet transmission delay requirements. In this paper, we analyze the effects of the throughput and strict safety packet transmission delay with adaptive multichannel MAC protocols, such as connectivity‐aware MAC (CA MAC), adaptive multi‐priority distributed MAC (APDM), multi‐priority supported p‐persistent MAC (MP MAC), and variable control channel interval MAC (VCI) protocols. The performance and analysis results show that: (a) under a low data rate condition, CA MAC does not guarantee a strict safety packet transmission delay; (b) APDM not only satisfies the safety packet transmission requirement, but also provides the lowest safety packet transmission delay; (c) under a high data rate condition, we suggest APDM for use as an adaptive MAC protocol because it allows for high throughput for nonsafety packets and preserves low safety packet transmission delay; (d) under a low data rate condition with various data packet sizes, we suggest MP MAC for high throughput, which satisfies the safety packet transmission requirement; and (e) under low vehicle density and low data rate conditions, VCI can support high throughput. A balance between transmission delay and throughput must be considered to improve the optimal efficiency, reliability, and adaptability.     

DCNC: throughput maximization via delay controlled network coding for wireless mesh networks

Network coding (NC) can greatly improve the performance of wireless mesh networks (WMNs) in terms of throughput and reliability, and so on. However, NC generally performs a batch‐based transmission scheme, the main drawback of this scheme is the inevitable increase in average packet delay, that is, a large batch size may achieve higher throughput but also induce larger average packet delay. In this work, we put our focus on the tradeoff between the average throughput and packet delay; in particular, our ultimate goal is to maximize the throughput for real‐time traffic under the premise of diversified and time‐varying delay requirements. To tackle this problem, we propose DCNC, a delay controlled network coding protocol, which can improve the throughput for real‐time traffic by dynamically controlling the delay in WMNs. To define an appropriate control foundation, we first build up a delay prediction model to capture the relationship between the average packet delay and the encoding batch size. Then, we design a novel freedom‐based feedback scheme to efficiently reflect the reception of receivers in a reliable way. Based on the predicted delay and current reception status, DCNC utilizes the continuous encoding batch size adjustment to control delay and further improve the throughput. Extensive simulations show that, when faced with the diversified and time‐varying delay requirements, DCNC can constantly fulfill the delay requirements, for example, achieving over 95% efficient packet delivery ratio (EPDR) in all instances under good channel quality, and also obtains higher throughput than the state‐of‐art protocol. Copyright © 2014 John Wiley & Sons, Ltd.     

Joint optimization and threshold structure dynamic programming with enhanced priority scheme for adaptive VANET MAC

Nodes in vehicular ad-hoc network (VANET) are highly mobile, traversing in unpredictable and varying environment. Therefore, contention window size and transmission power should adapt according to the high mobility transmission environment. In this paper, we propose an adaptive VANET medium access control (MAC) layer with joint optimization for VANET (MACVS) which aims at minimizing average delay and maximizing packet success rate. An adaptive joint optimization with proposed threshold structure dynamic programming, with closed loop feedback control system, is designed to optimize contention window size and transmission power. Adaptive optimization is done based on road traffic conditions and transmission reliable distance range (depicted by interference and noise), by monitoring the continuous change and threshold of received signal strength to interference and noise ratio. Mathematical expressions have been developed for the MACVS optimization framework, and the produced analytical results show good agreement with the simulation results. Simulations with different arrival rates and urban map of city center show that the proposed MACVS with low complexity joint optimization effectively reduces end-to-end delay while achieving high packet success rate under various network traffic condition.     

Design and Implementation of a QoS Oriented Data-Link Control Protocol for CBR Traffic in Wireless ATM Networks

This paper presents a QoS oriented Data Link Control (DLC) framework for transporting Constant Bit Rate (CBR) traffic over wireless ATM links. Data link control is usually omitted in fixed ATM networks because cell corruption due to channel error is extremely rare for reliable media like copper wire and optical fiber. However, for wireless, higher bit error rates are quite common due to shadowing and other fading effects. The purpose of DLC in wireless is to provide error-free transport to the higher layers by recovering corrupted cells at the link layer. A selective reject (SREJ) automatic repeat request (ARQ) based DLC protocol is used for CBR error recovery. For an ARQ based scheme, higher recovery rates can be achieved with larger cell transfer delay, caused by cell retransmissions. Since cell transfer delay and DLC recovery rate both translate to user-perceivable Quality-of-Service (QoS), it is important for the DLC to strike a balance between these two, depending on the application's requirements. To achieve this in our protocol, the retransmission procedure for a CBR cell is constrained to complete within a recovery time interval which is specified by the application at call-setup time. Also, a novel jitter removal scheme that reduces the cell delay variation caused by cell loss and retransmissions, is incorporated as a part of the DLC protocol. The proposed protocol is implemented on NEC's WATMnet prototype system. The implementation and its experimental results are reported for illustrating the performance and feasibility of the presented CBR DLC protocol. The experimental results show that the DLC protocol can be successfully applied for QoS-constrained error recovery of CBR traffic on a per-connection basis. These also indicate that the DLC can be programmed to attain a desirable tradeoff between cell transfer delay and cell recovery rate.     

A Modified Dynamic Weighted Round Robin Cell Scheduling Algorithm

In this paper, we propose the modified dynamic weighted round robin (MDWRR) cell scheduling algorithm, which guarantees the delay property of real‐time traffic and also efficiently transmits non‐real‐time traffic. The proposed scheduling algorithm is a variation of the dynamic weighted round robin (DWRR) algorithm and guarantees the delay property of real‐time traffic by adding a cell transmission procedure based on delay priority. It also uses a threshold to prevent the cell loss of non‐real‐time traffic that is due to the cell transmission procedure based on delay priority. Though the MDWRR scheduling algorithm may be more complex than the conventional DWRR scheme, considering delay priority minimizes cell delay and decreases the required size of the temporary buffer. The results of our performance study show that the proposed scheduling algorithm has better performance than the conventional DWRR scheme because of the delay guarantee of real‐time traffic.     

Adaptive Input Traffic Prediction Scheme for Absolute and Proportional Delay Differentiated Services in Broadband Convergence Network

In this paper, an algorithm that provides absolute and proportional differentiation of packet delays is proposed with the objective of enhancing quality of service in future packet networks. It features an adaptive scheme that adjusts the target delay for every time slot to compensate the deviation from the target delay, which is caused by prediction error on the traffic to arrive at the next time slot. It predicts the traffic to arrive at the beginning of a time slot and measures the actual arrived traffic at the end of the time slot. The difference between them is utilized by the delay control operation for the next time slot to offset it. Because the proposed algorithm compensates the prediction error continuously, it shows superior adaptability to bursty traffic and exponential traffic. Through simulations we demonstrate that the algorithm meets the quantitative delay bounds and is robust to traffic fluctuation in comparison with the conventional non‐adaptive mechanism. The algorithm is implemented with VHDL on a Xilinx Spartan XC3S1500 FPGA, and the performance is verified under the test board based on the XPC860P CPU.     

Rate Proportional SCFQ Algorithm for High‐Speed Packet‐Switched Networks

Self‐Clocked Fair Queueing (SCFQ) algorithm has been considered as an attractive packet scheduling algorithm because of its implementation simplicity, but it has unbounded delay property in some input traffic conditions. In this paper, we propose a Rate Proportional SCFQ (RP‐SCFQ) algorithm which is a rate proportional version of SCFQ. If any fair queueing algorithm can be categorized into the rate proportional class and input is constrained by a leaky bucket, its delay is bounded and the same as that of Weighted Fair Queueing (WFQ) which is known as an optimal fair queueing algorithm. RP‐SCFQ calculates the timestamps of packets arriving during the transmission of a packet using the current value of system potential updated at every packet departing instant and uses a starting potential when it updates the system potential. By doing so, RP‐SCFQ can have the rate proportional property. RP‐SCFQ is appropriate for high‐speed packet‐switched networks since its implementation complexity is low while it guarantees the bounded delay even in the worst‐case input traffic conditions.