基于无线传感器网络的跨层拥塞控制协议

无线传感器网络(WSN)中由拥塞引起的大量分组重传以及重传多次失败后的分组丢弃会导致较长的时延、较高的分组丢失率和较多的能量消耗.为了准确探测和控制网络拥塞,提出了一种基于跨层设计的拥塞控制协议,即上行拥塞控制(UCC)协议.该协议利用节点在媒质接人控制(MAC)层中未占用的缓冲器区间大小和所预测的通信流量作为该节点的...     

Upstream congestion control in wireless sensor networks through cross-layer optimization

Congestion in wireless sensor networks not only causes packet loss, but also leads to excessive energy consumption. Therefore congestion in WSNs needs to be controlled in order to prolong system lifetime. In addition, this is also necessary to improve fairness and provide better quality of service (QoS), which is required by multimedia applications in wireless multimedia sensor networks. In this paper, we propose a novel upstream congestion control protocol for WSNs, called priority-based congestion control protocol (PCCP). Unlike existing work, PCCP innovatively measures congestion degree as the ratio of packet inter-arrival time along over packet service time. PCCP still introduced node priority index to reflect the importance of each sensor node. Based on the introduced congestion degree and node priority index, PCCP utilizes a cross-layer optimization and imposes a hop-by-hop approach to control congestion. We have demonstrated that PCCP achieves efficient congestion control and flexible weighted fairness for both single-path and multi-path routing, as a result this leads to higher energy efficiency and better QoS in terms of both packet loss rate and delay.     

Mitigation of Packet Loss Using Data Rate Adaptation Scheme in MANETs

Node’s mobility, bursty data traffic, and dynamic nature of the network make congestion avoidance and control a challenging task in Mobile Adhoc Networks (MANETs). Congestion results in high packet loss rate, increased delays, and wastage of network resources due to re-transmissions. In this paper, we propose In-route data rate adaptation to avoid packet loss. Proposed scheme is based on the analysis of queue length of the forwarding nodes, number of data source nodes, and rate of link changes. In proposed technique, queue length of forwarding nodes is communicated periodically to the neighbor nodes using existing control messages of the underlying routing protocol. Keeping in view the queue length of forwarding nodes, number of data source nodes, and rate of link changes, initially the intermediate nodes buffer the incoming data packets upto some threshold and then, gradually shift the effect of congestion to the data source nodes. Then, the source node adapts its sending data rate to avoid congestion and to ensure reliable data communication. We have performed simulations in NS-2 simulator by varying different network metrics such as data rate, number of source nodes, and node speed. Results show that proposed technique improves network performance in terms of packet delivery ratio upto 15 %, reduction of average end-to-end delay and packet loss due to interface queue overflow upto 25 % and 14 % respectively, as compared to the static rate adaptation scheme.     

一种新的基于虚拟队列的无线多播网络编码调度策略

网络编码由于其传输效率高的特性,近年来在无线多播网络中得到广泛的应用。针对无线多播网络中丢包自动重传效率低的问题,该文提出一种新的基于虚拟队列中数据包到达时间的编码调度策略(CSAT)。在CSAT策略中,为了提高编码效率,采用虚拟队列来存放初始以及未被所有接收者接收到的数据包。考虑到队列的稳定性,CSAT策略按照一定的比率从主次队列选择发送;在次队列发送数据包时,结合了编码和非编码两种方式,根据数据包到达队列的先后,选取能够使较多数据包参与编码的方式发送。仿真结果表明,该文所提的CSAT编码调度策略在有效提高了数据包传输效率的同时,提高了网络的吞吐量并降低了平均等待时延。     

Energy Efficient Reliable Multi-path Data Transmission in WSN for Healthcare Application

In healthcare applications of WSN, the data loss due to congestion may cause death alarm for a patient in critical condition. Therefore, an efficient congestion avoidance or otherwise an efficient congestion control mechanism is required. In this paper, we present an energy efficient reliable multi-path data transmission protocol for reliable data transport over WSN for the health care application. The emergency data and sensitive data packets are transmitted through an alternate path having minimum correlation with transmission interference during congestion. The proposed protocol attempts to avoid congestion by computing the probability of congestion at the intermediate nodes and transmission rate at the intermediate node is adjusted. The buffer of each node is partitioned to support fair and efficient data delivery. The reliability of the proposed protocol is achieved through hop-by-hop loss recovery and acknowledgement. The performance of the proposed protocol is evaluated through extensive simulations. The simulation results reveal that it outperforms the existing congestion control protocols for healthcare application in terms of energy efficiency, reliability and end-to-end delivery ratio.     

Queue‐based multiple path load balancing routing protocol for MANETs

Congestion in the network is the main cause for packet drop and increased end‐to‐end transmission delay of packet between source and destination nodes. Congestion occurs because of the simultaneous contention for network resources. It is very important to efficiently utilize the available resources so that a load can be distributed efficiently throughout the network. Otherwise, the resources of heavily loaded nodes may be depleted very soon, which ultimately affects network performances. In this paper, we have proposed a new routing protocol named queue‐based multiple path load balancing routing protocol. This protocol discovers several node‐disjoint paths from source to destination nodes. It also finds minimum queue length with respect to individual paths, sorts the node‐disjoint paths based on queue length, and distributes the packets through these paths based on the minimum queue length. Simulation results show that the proposed routing protocol distributes the load efficiently and achieves better network performances in terms of packet delivery ratio, end‐to‐end delay, and routing overhead. Copyright © 2016 John Wiley & Sons, Ltd.     

Packet sequencing: a deterministic protocol for QoS in IP networks

We describe a deterministic protocol for routing delay and loss-sensitive traffic through an IP network. Unlike traditional approaches, the method described here - packet sequencing - does not rely on queue management. Instead, it uses a temporally-based deterministic protocol to coordinate and switch IP packets on a systemwide basis. As a result, end-to-end throughput is guaranteed, without packet loss, loss variance, or accumulated performance impairment; additionally, end-to-end delay is minimized, and jitter is essentially eliminated. We also show that packet sequencing can complement conventional IP networks: sequencing does not negate the use of queue management QoS methods that are the subject of considerable ongoing study. This article describes the fundamental approach, issues associated with scalability, illustrative performance in the context of storage networking, and attributes related to the security and reliability of IP networks.     

一种基于路径信息的WSN路由协议

AODV是一种广泛应用于MANET的按需路由协议,但是它并不完全适用于WSN。本文通过改进AODV协议基于洪泛的路由机制,提出了一种适用于WSN的低开销、低时延的路由协议IAODV。仿真结果表明:与AODV协议相比,IAODV协议可以在保持较高的分组投递率的同时有效降低数据分组的平均端到端时延和路由开销,达到了低开销、低时延的设计目标。     

A Delay-Aware Congestion Control Protocol for Wireless Sensor Networks

In wireless sensor networks, congestion leads to buffer overflowing, and increases delay. The tradi-tional solutions use rate adjustment to mitigate congestion, thus increasing the delay. A Delay-aware congestion con-trol protocol (DACC) was presented to mitigate congestion and decrease delay. In order to improve the accuracy of the existing congestion detection model which is based on the buffer occupancy of a single node, DACC presents a new model considering both the real-time buffer occupancy and the average transmission time of packets. DACC uses the untapped bits in the IEEE 802.11 Distributed coordination function (DCF) frames header to carry congestion infor-mation. During the congestion alleviation period, DACC presents a channel occupancy mechanism which is based on the real-time buffer occupancy for the purpose of decreas-ing delay and preventing packet loss. Simulation results indicate that in terms of delay, packet delivery ratio, col-lision and buffer load, DACC has comparative advantages than those of 802.11 DCF, Priority-based congestion con-trol protocol (PCCP) and Decoupling congestion control and fairness (DCCF).     

Proposing a Method for Controlling Congestion in Wireless Sensor Networks Using Comparative Fuzzy Logic

Recent developments and advances on electronics and wireless telecommunications have enabled researchers to design and produce low-power and small sensors with reasonable prices which can be used for various applications. Wireless multimedia sensor networks are a new subset of WSN family which is capable of doing operations such as receiving multimedia information, i.e. video, sound, photo and numerical data from the surrounding environment, processing them and transmitting them. Due to high transmission rate and the explosive feature, the transmission of video flows in WSNs concerned with several challenges. Congestion also leads to the loss of packets and costly retransmission of packets. Consequently, the limited energy of the sensor nodes is wasted. Accordingly, in this paper, using fuzzy logic, a new congestion control method was proposed for these networks. In the proposed method, congestion announcement and control are carried out by using three main parameters, i.e. the remaining energy level of the node, load density and accessible detection bandwidth. The results of evaluations, done via OPNET 11.5, indicated that using the proposed method led to an average delay reduction in packet arrival. Also, less energy of the nodes is consumed and network lifetime is enhanced. Multimedia is used for novel approaches such as the followings: communications, commerce, education, entertainment, personal locator services, advanced health care, control systems, traffic avoidance and execution and in Information Technology.     

TCP-DCR: a novel protocol for tolerating wireless channel errors

This paper presents TCP-DCR, a set of simple modifications to the TCP protocol to improve its robustness to channel errors in wireless networks. TCP-DCR is based on the simple idea of allowing the link-level mechanism to recover the packets lost, due to channel errors, thereby limiting the response of the transport protocol to mostly congestion losses. This is done by delaying the triggering of congestion response algorithms for a small bounded period of time /spl tau/ to allow the link-level retransmissions to recover the loss due to channel errors. If at the end of the delay /spl tau/ the packet is not recovered, then it is treated as a packet lost due to congestion. We analyze TCP-DCR to show that the delay in congestion response does not impact the fairness towards the native implementations of TCP that respond to congestion immediately after receiving three dupacks. We evaluate TCP-DCR through simulations to show that it offers significantly better performance when channel errors contribute more towards packet losses in the network with no or minimal impact on the performance when congestion is the primary cause for packet loss. We also present an analysis to show that the number of flows in the network significantly influences protocol evaluation in the wireless networks.     

WSN中基于势能场的多策略路由协议

文章结合无线传感器网络（WSN）中流量分布的向心性特点,借鉴物理学中势能场的概念与机理,提出一种开放的路由协议实现框架。利用不同的网络参数构造不同的“虚拟势能场”,叠加后形成的复合势能场将驱动数据分组沿着势场梯度变化最快的方向移动,一方面可以最终将网络中的数据分组汇聚于目的节点,实现路由协议的基本功能;同时,在动态时变“虚拟势能场”的调节下,还可以为路由协议附加各种有利于实现其他优化目标的策略与属性,如能耗均衡、拥塞避免、支持服务质量保障和利于数据聚合等,在无线传感器网络中实现多策略路由。作为例子,文章提出了一个基于势能场的提供实时传输的路由协议,它能在严格保障实时分组获得最小化端到端延迟的前提下,有效缓解网络拥塞,提高全网吞吐量。     

Congestion control for packet voice by selective packet discarding

In order to reduce the time delays as well as multiplexer memory requirements in packet voice systems, a family of congestion control schemes is proposed. They are all based on the selective discarding of packets whose loss will produce the least degradation in quality of the reconstructed voice signal. A mathematical model of the system is analyzed and queue length distributions are derived. These are used to compute performance measures, including mean waiting time and fractional packet loss. Performance curves for some typical systems are presented, and it is shown that the control procedures can achieve significant improvement over uncontrolled systems, reducing the mean waiting time and total packet loss (at transmitting and receiving ends). Congestion control with a resume level is also analyzed, showing that without increasing the fractional packet loss, the mean and variance of the queue can be reduced by selecting an appropriate resume level. The performance improvements are confirmed by the results of some informal subjective testing     

Dynamic switching based real-time routing in low-duty-cycle wireless sensor networks

The dynamic switching based real-time(DSRT)routing protocol was proposed to handle the arbitrary end-to-end(E2E)real-time communication in the low-duty-cycle wireless sensor networks.Firstly,the concept of available speed was designed to compensate for the big sleep latency and facilitate discovering the routes with less latency based on two-hop neighbors’ information(at lease about 20% routing path with less latency was discovered by DRST in the experiments).Moreover,it was noticed that the congestion extent in the low-duty-cycle network was determined not only by the number of packets in the network output queue,but also the destination of the packets.However,the traditional method with one-hop neighbors’ information cannot differentiate this kind of congestion.Therefore,combined with the dynamic switching mechanism,the DSRT proposed a congestion avoiding algorithm by classifying the packets in the queue.Through comprehensive experiments,the efficiency of routing discovering and congestion avoiding of the DSRT protocol is demonstrated,and the E2E delay is decreased by at least 200% when the traffic was high.     

A novel adaptive traffic prediction AQM algorithm

In the Internet, network congestion is becoming an intractable problem. Congestion results in longer delay, drastic jitter and excessive packet losses. As a result, quality of service (QoS) of networks deteriorates, and then the quality of experience (QoE) perceived by end users will not be satisfied. As a powerful supplement of transport layer (i.e. TCP) congestion control, active queue management (AQM) compensates the deficiency of TCP in congestion control. In this paper, a novel adaptive traffic prediction AQM (ATPAQM) algorithm is proposed. ATPAQM operates in two granularities. In coarse granularity, on one hand, it adopts an improved Kalman filtering model to predict traffic; on the other hand, it calculates average packet loss ratio (PLR) every prediction interval. In fine granularity, upon receiving a packet, it regulates packet dropping probability according to the calculated average PLR. Simulation results show that ATPAQM algorithm outperforms other algorithms in queue stability, packet loss ratio and link utilization.     

Distributed collision control with the integration of packet size for congestion control in wireless sensor networks

Several great features offered by wireless sensor networks (WSN) result in its wide deployment in various remote and continuous monitoring applications. As such, managing huge collected readings in this domain posted many challenges due to its design limitations. In order to provide seamless data transmission, which is of utmost importance in those delay‐sensitive applications, minimum delay and packet loss occurrence should be considered. Specifically, this paper addresses the common issue of congested networks in WSN with the combination technique of variance‐based distributed contention control (DCC‐V) and packet size optimization. The proposed integration technique, which operates on medium access control layer, takes into consideration the packet size advantages as it plays a key role in determining successful data delivery, given the error‐prone nature of WSN. While ensuring fewer corrupted packets, the proposed contention window (CW) in DCC‐V minimizes the chances of packet collisions and so alleviates congestion. In this technique, CW is determined based on slot utilization and average collision values, which also involve standard deviation measurements. Simulation analysis using network simulator‐2 shows outstanding performance of the proposed solution compared with the existing IEEE 802.15.4 protocol. Copyright © 2014 John Wiley & Sons, Ltd.     

Multipath interconnection: a technique for reducing congestionwithin fast packet switching fabrics

The banyan interconnection is prone to internal link congestion, resulting in a blocking switch architecture. Several solutions that have been implemented to reduce the severity of link congestion offer packets a multiplicity of paths, which tend to increase packet delay variability and allow delivery of out-of-sequence packets. This, in turn, can lead to an increase in end-to-end protocol complexity, particularly in the case of real-time services. A solution called multipath interconnection is proposed to overcome this difficulty. Multiple (i.e., alternate) paths are provided and one is selected at call-setup time. Subsequent packets belonging to the call are constrained to follow the selected path. A number of path selection strategies are presented     

Congestion control in interconnected LANs

The reasons why congestion control is more difficult in interconnected local area networks (LANs) than in conventional packet nets are examined. The flow and congestion control mechanisms that can be used in an interconnected LAN environment are reviewed. The focus is on congestion control (that is, prevention of internal congestion); however some of the proposed schemes require the interaction of flow and congestion control. The schemes considered are dropping packets; input buffer limit, i.e. a limit on the number of input packets (i.e. packets from local hosts) that can be buffered in the packet switch; the use of choke packets, in which, whenever a bridge or router experiences congestion, it returns to the source a choke packet containing the header of the packet traveling in the congested direction and the source, on receiving the choke packet, declares the destination congested, and slows (or stops altogether, for a period of time) traffic to that destination; backpressure, which is the regulation of flow along a virtual connection; and congestion prevention, whereby a voice or video connection is accepted only if there is enough bandwidth (in a statistical sense) in the network to support it     

Evaluation of the effects of protocol processing overhead in errorrecovery schemes for a high-speed packet switched network: link-by-linkversus edge-to-edge schemes

The authors investigate the effects of protocol processing overhead on the performance of error recovery schemes. The focus is on the edge-to-edge error recovery scheme, in which retransmissions of erred packets only take place between source and destination nodes. An approximation is obtained for the Laplace transform for the distribution of the end-to-end packet transfer delay, considering the processing time required for error recovery. The performance of the link-by-link error recovery scheme, in which retransmissions take place between adjacent nodes, is evaluated and compared to the performance of the edge-to-edge scheme. Numerical results from a tandem queuing network model show that for a network with very-high-speed/low-error-rate channels, an edge-to-edge scheme gives a smaller packet transmission delay than a link-by-link scheme for both go-back-N and selective-repeat retransmission procedures, while keeping the packet loss probability sufficiently small     

A reliable overlay video transport protocol for multicast agents in wireless mesh networks

A new video transport protocol for multicast agents in wireless mesh networks (WMNs) is proposed in this paper. The proposed protocol enables a significant reduction in the transmission overhead, while providing reliable communication for its use in multicast applications. This proposed reliable protocol provides a practical approach for an overlay peer‐to‐peer multicast facility supported within the application layer. This obviates the need to give upgraded routers capable of handling multicast broadcasting or modify the existing protocol stack. The protocol tolerates partial losses in multimedia transmissions, while supporting control of the delay sensitivity of such transmissions in WMNs. The key issue in this protocol is the ability to detect packet loss, anticipate retransmission requests, and use the anticipated retransmission requests to transmit the lost packets prior to requests from other receiving agents. The proposed protocol allows for the receiver to determine if retransmission of lost packets is required, ensuring the greatest flexibility needed for a reliable multicast protocol. Copyright © 2009 John Wiley & Sons, Ltd.