A request control scheme for data recovery in DVB‐IPDC systems with spatial and temporal packet loss

Recently, digital video broadcasting‐handheld (DVB‐H) and IP datacast over DVB‐H (DVB‐IPDC) have been developed to support broadcasting services. DVB‐H is designed to support digital video broadcast for handheld devices, whereas DVB‐IPDC can integrate with an IP‐relay network to complement the data loss problem in DVB‐H. Assuming that WiMAX networks are adopted to support DVB‐IPDC, this paper points out two critical problems: group packet loss (GPL) and broadcast data handover (BDH). GPL occurs when there is a burst of retransmission requests for the same pieces of data with high spatial or temporal correlation. BDH happens when some devices that made the above requests handover to new serving cells. To solve these problems, we propose lazy wait and group acknowledgement schemes to alleviate duplicate requests by exploiting their spatial and temporal correlations. This not only reduces the requests submitted by neighboring devices in both space and time domains but also avoids handovering devices from sending duplicate requests in new cells. Through mathematical analysis, we show how to adaptively adjust the timers of lazy wait and group acknowledgement based on channel quality. Simulation results prove that our schemes can efficiently reduce retransmission requests and retransmission packets, thus alleviating congestion in the IP‐relay network. Copyright © 2011 John Wiley & Sons, Ltd.     

To‐send‐or‐not‐to‐send: An optimal stopping approach to network coding in multi‐hop wireless networks

Network coding is all about combining a variety of packets and forwarding as much packets as possible in each transmission operation. The network coding technique improves the throughput efficiency of multi‐hop wireless networks by taking advantage of the broadcast nature of wireless channels. However, there are some scenarios where the coding cannot be exploited due to the stochastic nature of the packet arrival process in the network. In these cases, the coding node faces 2 critical choices: forwarding the packet towards the destination without coding, thereby sacrificing the advantage of network coding, or waiting for a while until a coding opportunity arises for the packets. Current research works have addressed this challenge for the case of a simple and restricted scheme called reverse carpooling where it is assumed that 2 flows with opposite directions arrive at the coding node. In this paper, the issue is explored in a general sense based on the COPE architecture requiring no assumption about flows in multi‐hop wireless networks. In particular, we address this sequential decision making problem by using the solid framework of optimal stopping theory and derive the optimal stopping rule for the coding node to choose the optimal action to take, ie, to wait for more coding opportunity or to stop immediately (and send packet). Our simulation results validate the effectiveness of the derived optimal stopping rule and show that the proposed scheme outperforms existing methods in terms of network throughput and energy consumption.     

基于机会式网络编码的低时延广播传输算法

为了提高无线网络中数据包广播传输的效率,本文提出了一种基于机会式网络编码的广播传输算法.该算法在发送端按一定顺序选择不同终端的丢包,并采用异或运算编码重传包,在终端采用从重传包中解码数据包的方法恢复丢包.该算法优先恢复时间重要性较高的丢包,并使多个终端同时从单个重传包恢复其丢包,因此有效地提高了广播传输效率并降低了传输...     

无线网络中基于网络编码的可靠广播传输

Recently, network coding has been applied to the loss recovery of reliable broadcast transmission in wireless networks. Since it was proved that fi nding the optimal set of lost packets for XOR-ing is a complex NP-complete problem, the available time-based retransmission scheme and its enhanced retransmission scheme have exponential computational complexity and thus are not scalable to large networks. In this paper, we present an efficient heuristic scheme based on hypergraph coloring and also its enhanced ...     

联合蜂窝与D2D链路的网络编码广播重传方案

移动设备之间的合作以及利用蜂窝和D2D链路等多个接口有望满足日益增长的吞吐量需求。考虑设备配备有双接口的无线D2D广播（Network coding for dual interfaces,NCDI）场景,重传阶段,设备同时利用蜂窝与D2D链路来恢复丢失数据包。然而,如何合理的进行编码调度,充分发挥网络编码增益显得至关重要。为最小化重传次数,文章旨在设计联合蜂窝与D2D链路的网络编码广播重传方案。针对随机线性网络编码（RLNC）与立即可译网络编码（IDNC）,分别提出了NCDI-RLNC以及NCDI-IDNC方案。仿真结果表明,与其他方案相比,提出的两种方案均能够有效地提高重传效率、减少重传次数。      

Swarm intelligence‐based anycast routing protocol in ubiquitous networks

As digital devices with communication capability become more pervasive, we are entering the era of ubiquitous computing, as predicted by Mark Weiser. In ubiquitous environments, distributed context management servers are deployed everywhere to provide information and computing resources for users anytime and anywhere. Smart handheld computing devices with context‐aware applications may retrieve context information from the nearest server. This study investigates the problem of routing packets to the nearest server in a ubiquitous environment. An anycast routing protocol based on swarm intelligence, referred to as ARPSI, is proposed to route packets dynamically to a nearby server in a mobile, ad hoc, wireless network. Based on swarm intelligence, ARPSI is able to find a short path to a neighboring server efficiently and quickly. Simulations are conducted to evaluate the performance of ARPSI and our simulation results show that ARPSI achieves a higher packet delivery ratio, shorter routing path to anycast servers, and lower control packet overhead than the AODV‐based anycast protocol (A‐AODV) protocol. Copyright © 2009 John Wiley & Sons, Ltd.     

无线网络中基于混合流的高效可靠机会网络编码算法

Although the wireless network is widely used in many fields, its characteristics such as high bit error rate and broadcast links may block its development. Network coding is an artistic way to exploit its intrinsic characteristics to increase the network reliability. Some people research network coding schemes for inter flow or intra flow, each type with its own advantages and disadvantages. In this paper, we propose a new mechanism, called MM NCOPE, which integrates the idea of inter flow and intra flow coding. On the one hand, MM NCOPE utilizes random liner coding to encode the NCOPE packets while NCOPE is a sub protocol for optimizing the COPE algorithm by iteration. In NCOPE, packets are automatically matched by size to be coded. As a result, it improves the coding gain in some level. On the other hand, we adopt the partial Acknowledgement retransmission scheme to achieve high compactness and robustness. ACK is an independent packet with the highest priority rather than a part of the data packets. Compared with existing works on opportunistic network coding, our approach ensures the reliability of wireless links and improves the coding gain.     

Downlink media streaming with wireless fountain coding in wireline‐cum‐WiFi networks

This paper addresses the problem of streaming packetized media data in a combined wireline/802.11 network. Since the wireless channel is normally the bottleneck for media streaming in such a network, we propose that wireless fountain coding (WFC) be used over the wireless downlink in order to efficiently utilize the wireless bandwidth and exploit the broadcast nature of the channel. Forward error correction (FEC) is also used to combat errors at the application‐layer. We analytically obtain the moment generating function (MGF) for the wireless link‐layer delay incurred by WFC. With the MGF, the expected value of this wireless link‐layer delay is found and used by the access point (AP), who has no knowledge of the buffer contents of wireless receivers, to make a coding‐based decision. We then derive the end‐to‐end packet loss/late probability based on the MGF. We develop an integrated ns‐3/EvalVid simulator to evaluate our proposed system and compare it with the traditional 802.11e scheme which is without WFC capability but equipped with application‐ and link‐layer retransmission mechanisms. Through extensive simulations of video streaming, we show that streaming with WFC is able to support more concurrent video flows compared to the traditional scheme. When the deadlines imposed on video packets are relatively stringent, streaming with WFC also shows superior performance in terms of packet loss/late probability, video distortion, and video frame delay, over the traditional scheme. Copyright © 2011 John Wiley & Sons, Ltd.     

IPv6 networks over DVB‐RCS satellite systems

Satellite plays an important role in global information infrastructure (GII) and next generation networks (NGNs). Similarly, satellite communication systems have great advantages to support IPv6 (Internet Protocol version 6) networks as a technology that allows universal access to broadband e‐services (audio, video, VPN, etc.). In the context of DVB‐S2 (digital video broadcast‐satellite) and DVB‐RCS (digital video broadcast‐return channel via satellite) standards, this paper presents the current SatSix project (satellite‐based communications systems within IPv6 networks) within the European 6th Framework Programme, which is implementing innovative concepts and effective solutions (in relation with the economical cost) for broadband satellite systems and services using the technology presented above. This project is promoting the introduction of the IPv6 protocol into satellite‐based communication systems. Moreover, through SatSix, the industry is addressing the next generation Internet, IPv6. It also enhances its competitive position in satellite broadband multimedia systems by exploiting the common components defined by the European DVB‐S2 and DVB‐RCS satellite broadband standards. Copyright © 2007 John Wiley & Sons, Ltd.     

High efficient multipacket decoding approach for network coding in wireless networks

To improve the performance of transmission by reducing the number of transmission and network overhead of wireless single-hop networks, this paper presents a high efficient multipacket decoding approach for network coding (EMDNC) in wireless networks according to the idea of encoding packets which cannot be decoded and are stored in buffer by receiving nodes, the lost packets can be recovered from these encoded packets. Compared with the network coding wireless broadcasting retransmission (NCWBR), EMDNC can improve the efficiency of decoding and reduce the number of retransmission and transmission delay. Simulation results reveal that EMDNC can effectively reduce the number of retransmission and network overhead.     

Fair coding for inter‐session network coding in wireless mesh networks

Because of the broadcast and overhearing capability of wireless networks, network coding can greatly improve throughput in wireless networks. However, our investigation of existing inter‐session network coding protocols found that the short‐term unfairness that existed in 802.11‐based medium access control (MAC) protocols actually decreases the coding opportunity, which in turn compromises the throughput gain of network coding. To alleviate the negative impact of this unfairness, we propose a coding‐aware cross‐layer heuristic approach to optimize the coordination of network coding and MAC layer protocol, named FairCoding, which can significantly increase coding opportunities for inter‐session network coding through a fair short‐term traffic allocation for different coding flows. Experiment evaluation shows that the proposed mechanism can bring more coding opportunities and improve the total throughput of wireless mesh networks by up to 20%, compared with the coding mechanism, without considering the negative impact of the short‐term unfairness. Copyright © 2015 John Wiley & Sons, Ltd.     

D2D网络中基于立即可解网络编码的时延最小化重传方案

该文针对D2D无线网络中多终端并发协作重传冲突避免问题,提出一种基于立即可解网络编码的时延最小化重传方案。在重传阶段,充分利用D2D无线网络终端协作传输数据的优势,结合各终端数据包接收状态,综合考虑时延的影响因素,选取单次重传时延增量较小的数据包生成编码包,最小化重传时延。同时,构建终端冲突图,在图中搜索极大独立集,根据各终端的编码包权重值,选择最大加权独立集中的终端作为并发协作重传终端,从而降低重传次数。仿真结果表明,所提方案能够进一步改善D2D无线网络的重传效率。     

Decode‐and‐forward two‐way relaying protocol with one retransmission

This paper investigates the decode‐and‐forward two‐way relaying channel without direct link and proposes a protocol based on the physical‐layer network coding (PNC) protocol. The proposed protocol (termed ORT) introduces one retransmission into PNC, aiming at enhancing its outage performance. To manifest the merits of ORT, we compare it with PNC and the time‐division broadcast (TDBC) protocol, in terms of outage performance, expected rate, and diversity‐multiplexing tradeoff (DMT). Firstly, we derive the outage probability of the three protocols and then the expected rate. Secondly, asymptotic analysis is conducted to shed light on the diversity and coding gains. Finally, the DMT is obtained for the three protocols. The numerical results reveal the following: (i) that ORT performs better than PNC in both outage and expected rate performance when the nodes transmit with different powers. However, it has the same DMT performance with PNC; (ii) that ORT possesses improved DMT performance over TDBC whereas its expected rate is only better than the latter at medium to high signal‐to‐noise ratio. Copyright © 2012 John Wiley & Sons, Ltd.     

Transport layer enhancements on a satellite‐based mobile broadband system for high‐speed trains

The great diffusion of wireless portable and handheld devices, along with the growing availability of applications on the Internet, led to a significant increase in the demand of access. Users require to continue business or private activities when moving, with performance at least comparable with that experienced over fixed networks. A very interesting scenario encompasses broadband and uninterrupted access over high‐speed trains. Developments of new telecommunication standards as well as innovative network architectures give the potential to fully match the aforementioned requirements. Nevertheless, there is a significant number of issues, especially related to system integration, mobility management, and propagation channel conditions, that need a careful analysis with the aim to support the design of real systems. This paper deals with this challenging scenario proposing a satellite‐based architecture in which DVB‐S2 and DVB‐RCS standards, MIPv6 at layer 3 (for handover management), and an enhanced TCP version at layer 4, named TCP Noordwijk, allow to achieve good performance. An NS‐2 enhanced simulator has supported the whole analysis. Results show the effectiveness of the proposed scheme against different modulation and coding configurations and compared with a selected set of TCP‐enhanced versions. Copyright © 2012 John Wiley & Sons, Ltd.     

Network‐coding‐based cache policy for loss recovery enhancement in reliable multicast

In local loss recovery schemes, a small number of recovery nodes distributed along the transmission paths save incoming packets temporarily in accordance with a specified cache policy and retransmit these packets if they subsequently receive a request message from a downstream receiver. To reduce the recovery latency, the cache policy should ensure that the recovery nodes are always able to satisfy the retransmission requests of the downstream receivers. However, owing to the limited cache size of the recovery nodes and the behavior of the cache policy, this cannot always be achieved, and thus some of the packets must be retransmitted by the sender. Accordingly, this paper develops a new network‐coding‐based cache policy, designated as network‐coding‐based FIFO (NCFIFO), which extends the caching time of the packets at the recovery nodes without dropping any of the incoming packets. As a result, the lost packets can be always recovered from the nearest recovery nodes and the recovery latency is significantly reduced. The loss recovery performance of the NCFIFO cache policy is compared with that of existing cache policies by performing a series of simulation experiments using both a uniform error model and a burst error model. The simulation results show that the NCFIFO cache policy not only achieves a better recovery performance than existing cache policies, but also provides a more effective solution for managing a small amount of cache size in environments characterized by a high packet arrival rate. Copyright © 2011 John Wiley & Sons, Ltd.     

Syndrome: a light‐weight approach to improving TCP performance in mobile wireless networks

It is well known that the performance of TCP deteriorates in a mobile wireless environment. This is due to the fact that although the majority of packet losses are results of transmission errors over the wireless links, TCP senders still take packet loss as an indication of congestion, and adjust their congestion windows according to the additive increase and multiplicative decrease (AIMD) algorithm. As a result, the throughput attained by TCP connections in the wireless environment is much less than it should be. The key problem that leads to the performance degradation is that TCP senders are unable to distinguish whether packet loss is a result of congestion in the wireline network or transmission errors on the wireless links. In this paper, we propose a light‐weight approach, called syndrome, to improving TCP performance in mobile wireless environments. In syndrome, the BS simply counts, for each TCP connection, the number of packets that it relays to the destination host so far, and attaches this number in the TCP header. Based on the combination of the TCP sequence number and the BS‐attached number and a solid theoretical base, the destination host will be able to tell where (on the wireline or wireless networks) packet loss (if any) occurs, and notify TCP senders (via explicit loss notification, ELN) to take appropriate actions. If packet loss is a result of transmission errors on the wireless link, the sender does not have to reduce its congestion window. Syndrome is grounded on a rigorous, analytic foundation, does not require the base station to buffer packets or keep an enormous amount of states, and can be easily incorporated into the current protocol stack as a software patch. Through simulation studies in ns‐2 (UCB, LBNL, VINT network simulator, http://www‐mash.cs.berkeley.edu/ns/ ), we also show that syndrome significantly improves the TCP performance in wireless environments and the performance gain is comparable to the heavy‐weight SNOOP approach (either with local retransmission or with ELN) that requires the base station to buffer, in the worst case, a window worth of packets or states. Copyright © 2001 John Wiley & Sons, Ltd.     

车联网下基于网络编码的高吞吐量多径路由算法

在车联网中,由于车辆节点间无线链路的频繁中断易导致数据重传,使整个车辆组网性能急剧下降。为此,面向车联网提出了一种增强吞吐量的多径路由算法,其核心是使车辆节点分簇算法支持网络编码,从而达到对乱序和丢失报文恢复的目的。该算法首先采用协同编码通信模型来实现分簇后同簇车辆节点互相协同的多路径传输,然后采用网络编码对源端车辆和中间车辆节点发送的数据进行线性编码操作,最后在目的端节点予以解码。在基于QualNet仿真平台上的实验结果表明,该算法能够有效地提升车联网中多径路由的吞吐量。     

一种基于网络编码的重传算法研究

无线网络中,节点发送的数据分组传输失败后,执行重传机制。传统的重传机制ARQ由于在一次重传中只能发送一个丢失的数据分组,因此传输效率比较低。利用网络编码技术和AQR重传机制,我们可以在重传中使用网络编码,广播发送由多个丢失数据分组编码得到的编码分组,从而提高重传效率。本文中我们提出一种将网络编码应用于多个发送方多个接收方(MSMR)无线网络中的算法RMBNC。理论推导和仿真分析验证了我们提出的算法的有效性。     

Modelling of wireless TCP for short‐lived flows

The transmission control protocol (TCP) is one of the most important Internet protocols. It provides reliable transport services between two end‐hosts. Since TCP performance affects overall network performance, many studies have been done to model TCP performance in the steady state. However, recent researches have shown that most TCP flows are short‐lived. Therefore, it is more meaningful to model TCP performance in relation to the initial stage of short‐lived flows. In addition, the next‐generation Internet will be an unified all‐IP network that includes both wireless and wired networks integrated together. In short, modelling short‐lived TCP flows in wireless networks constitutes an important axis of research. In this paper, we propose simple wireless TCP models for short‐lived flows that extend the existing analytical model proposed in [IEEE Commun. Lett. 2002; 6 (2):85–88]. In terms of wireless TCP, we categorized wireless TCP schemes into three types: end‐to‐end scheme, split connection scheme, and local retransmission scheme, which is similar to the classification proposed in [IEEE/ACM Trans. Networking 1997; 756–769]. To validate the proposed models, we performed ns‐2 simulations. The average differences between the session completion time calculated using the proposed model and the simulation result for three schemes are less than 9, 16, and 7 ms, respectively. Consequently, the proposed model provides a satisfactory means of modelling the TCP performance of short‐lived wireless TCP flows. Copyright © 2005 John Wiley & Sons, Ltd.     

中继协作无线网络中不完美反馈下基于网络编码的重传方案

无线广播网络传输过程中,目的节点反馈信息丢失或部分丢失导致发送节点不能了解目的节点的真实接收状态.为提高不完美反馈下无线网络的重传效率,本文提出中继协作无线网络中不完美反馈下基于网络编码的重传方案.本方案基于部分可观察马尔科夫决策过程对不完美反馈下的重传过程进行建模.发送节点根据系统观测状态和最大置信度更新系统估计状态,根据数据包发送顺序,优先选择最早丢失且能够恢复最多丢包的编码包重传.目的节点缓存不可解编码包以提升编解码机会.重传过程中源节点关注目的节点请求包需求,相同情况优先选择传输可靠性较高的中继节点,以提升传输有效性.仿真结果表明,在不完美反馈下相对于传统方案,本方案可有效提高重传效率.