Training overhead for decoding random linear network codes in wireless networks

We consider multicast communications from a single source to multiple destinations through a wireless network with unreliable links. Random linear network coding achieves the min-cut flow capacity; however, additional overhead is needed for end-to-end error protection and to communicate the network coding matrix to each destination. We present a joint coding and training scheme in which training bits are appended to each source packet, and the channel code is applied across both the training and data. This scheme allows each destination to decode jointly the network coding matrix along with the data without knowledge of the network topology. It also balances the reliability of communicating the network coding matrices with the reliability of data detection. The throughput for this scheme, accounting for overhead, is characterized as a function of the packet size, channel properties (error and erasure statistics), number of independent messages, and field size. We also compare the performance with that obtained by individual channel coding of training and data. Numerical results are presented for a grid network that illustrate the reduction in throughput due to overhead.     

A Network Coding Scheme to Improve Throughput for IEEE 802.11 WLAN

In IEEE 802.11 infrastructure wireless local area network (WLAN), the communication between any two nodes is relayed by an access point (AP), which becomes the bottleneck of WLAN and severely restricts the overall throughput. It is well known that network coding technique is able to greatly improve the throughput of wireless networks. But, the available coding schemes do not make full advantage of channel capacity due to the fact that they pick at most one packet from each data flow for coding and the picked packets may have a great difference in packet size, wasting some channel capacity. To remedy the problem, in this paper, we propose the coding scheme that combines multiple buffered packets in one flow into a larger packet for coding so that the packets participating in coding have close sizes. We formulate an integer programming problem to find the optimal packet coding, which is solved by an optimal algorithm with relative high time complexity together with a heuristic algorithm with relative low time complexity. Simulation results show that the proposed coding scheme is able to greatly improve the throughput of WLAN and the throughput gain increases with the growth of the number of coding flows.     

Contention-based MAC protocols with erasure coding for wireless data networks

Contention-based medium access control (MAC) protocol is a key component for the success of wireless data networks. Conventional random access protocols like ALOHA and Carrier Sense Multiple Access (CSMA) suffer from packet collision which leads to low throughput. Aimed at improving the throughput performance, we propose to integrate erasure coding with contention-based MAC protocols for recovering collided packets. To demonstrate the effectiveness of this approach, we focus on combining erasure coding with slotted ALOHA and slotted non-persistent CSMA in this paper. The performances of the resulting protocols are evaluated by both analytical model and simulation. Simulation results match very well with analytical results and show that the system throughput is increased for low to medium traffic loading. Packet loss ratio is also improved considerably with our scheme when the maximum number of packet retransmission times is limited. However, the delay for our scheme is higher due to the longer waiting time in our scheme for recovering collided packets. It is also shown that delay can be significantly reduced if we choose appropriate coding parameters though throughput will be sacrificed.     

Throughput analysis of two-hop wireless networks with network coding

A Two-hop Wireless Network (TWN) is the basic topology structure that provides network coding opportunity for improving throughput. Network coding on a homogeneous TWN, in which all the data flows have the same packet size and all the links have the same transmission rate, has been extensively investigated. In this paper, network coding on more practical heterogeneous TWNs, featured by various packet sizes and transmission rates, is studied. Based on the Markov model, the throughput of the proposed network coding scheme, together with the throughput gain, is derived, which matches the simulation results very well. Numerical analyses indicate that, encoding the packets with close size and close transmission rate and enlarging buffer size at the relay node help in improving the throughput gain.     

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。     

Packet combining in frequency-hop spread-spectrum communicationsystems

The performance of frequency-hop spread-spectrum (FHSS) communication systems using hybrid automatic-repeat-request (ARQ) can be improved by combining current and prior transmissions at the receiver. Two methods for combining packets in systems that employ interleaving and Reed-Solomon (RS) coding are presented and analyzed for the partial-band interference channel. These methods use majority logic combining at the codeword level to make retransmission decisions. Bounded distance errors-and-erasures decoding and erasure generation by means of Viterbi's ratio threshold test (RTT) are incorporated in the analysis. Results of the analysis show that, with comparable packet error probabilities, the packet-combining schemes provide significant gains in throughput when compared with systems that do not employ combining     

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.     

Random linear network coding with ladder-shaped global coding matrix for robust video transmission

Robust video multicast in erasure networks using network coding (NC) to reduce the impact of packet loss is studied in this paper. In our proposed solution, random linear network coding (RLNC) is adopted at intermediate nodes of the network. RLNC linearly combines a group of packets by randomly selecting weighting coefficients on a finite field, and the loss of an RLNC-coded packet is equivalent to the loss of one constraint in a linear system of equations required for RLNC decoding. Unless the global coding coefficient matrix, or simply called the global coding matrix (GCM), is of full rank, a receive node cannot reconstruct all source packets. To address this rank deficiency problem, we propose to construct a special-structured GCM, called the ladder-shaped GCM (LGCM), for layered H.264/SVC (scalable video coding) video multicast. The ladder shape of the sparse coding matrix is maintained throughout the RLNC process to achieve two objectives: (1) to enable partial decoding of a block; and (2) to provide unequal erasure protection for H.264/SVC priority layers. Furthermore, quality degradation is minimized by optimizing the amount of redundancy assigned to each layer, and graceful quality degradation is achieved by error concealment (EC). Simulation results are given to demonstrate the superior performance of the proposed RLNC–LGCM scheme over the traditional RLNC with a generalGCM.     

Video-aware opportunistic network coding over wireless networks

In this paper, we study video streaming over wireless networks with network coding capabilities. We build upon recent work, which demonstrated that network coding can increase throughput over a broadcast medium, by mixing packets from different flows into a single packet, thus increasing the information content per transmission. Our key insight is that, when the transmitted flows are video streams, network codes should be selected so as to maximize not only the network throughput but also the video quality. We propose video-aware opportunistic network coding schemes that take into account both the decodability of network codes by several receivers and the importance and deadlines of video packets. Simulation results show that our schemes significantly improve both video quality and throughput. This work is a first step towards content-aware network coding.     

A Cross-layer Dual Queue Approach for Improving TCP Fairness in Infrastructure WLANs

Fairness is one of the most important performance measures in IEEE 802.11 Wireless Local Area Networks (WLANs), where channel is accessed through competition. In this paper, we focus on the fairness problem between TCP uplink and downlink flows in infrastructure WLANs from the cross-layer perspective. First, we show that there exists a notable discrepancy between throughput of uplink flow and that of downlink flow, and discuss its root cause from the standpoint of different responses to TCP data packet drop and TCP ACK packet drop at the access point (AP) buffer. In order to mitigate this unfairness, we propose a dual queue scheme, which works in a cross-layer manner. It employs two separate queues at the AP, one for the data packets of downlink TCP flows and another for the ACK packets of uplink TCP flows, and selects these queues with appropriate probabilities so that TCP per-flow fairness is improved. Moreover, we analyze the behavior of the dual queue scheme and derive throughputs of uplink and downlink flows. Based on this analysis, we obtain the optimal queue selection probabilities for fairness. Extensive simulation results confirm that the proposed scheme is effective and useful in resolving the TCP unfairness problem without deteriorating overall utilization.     

Counteracting malicious adversaries via secret and reliable coding mechanism in random network coding

This paper investigates the problem of constructing a secret and reliable network coding communication scheme against an adversary that can reselect the μ tapping links in different time slices and inject z erroneous packets into the network, while the network suffers from ρ packet erasures. We first derive a necessary condition for keeping the transmitted information secret from the adversary, while the network is only subject to eavesdropping attack, and no errors are injected. We then provide two network coding schemes for error and erasure correction, which can decode the transmitted information correctly. With that, a secret and reliable network coding is proposed by combining two error and erasure correction schemes with secret communication. We show that under the requirement of 2z + ρ+ μ < k ?2z + 2ρ communication can achieve a rate of k ?μ?z ?ρ packets, where k denotes the network capacity. Moreover, it ensures that the communicated information has complete reliability and information‐theoretic security from the adversary. In particular, the requirement of packet length does not need to be sufficiently large, which is close to the practical packet length. Finally, we also compare our scheme's performance with existing work; the numeric results and security analyses illustrate the performance and security of our scheme. Copyright © 2011 John Wiley & Sons, Ltd.     

Loss probabilities for messages with redundant packets feeding afinite buffer

The purpose of this paper is to obtain the distribution of the number of lost packets within a sequence of n consecutive packet arrivals into a finite buffer M/M/1 queue. We obtain explicit expressions for the multidimensional generating function of these probabilities based on a recursive scheme introduced by Cidon et al. (1993). We then analyze the loss probabilities of a whole message, and analyze the effect of adding redundant packets. We show that in both heavy traffic as well as in light traffic conditions, adding redundant packets results in decreasing the message loss probabilities     

Analysis of packet loss processes in high-speed networks

The packet loss process in a single-server queueing system with a finite buffer capacity is analyzed. The model used addresses the packet loss probabilities for packets within a block of a consecutive sequence of packets. An analytical approach is presented that yields efficient recursions for the computation of the distribution of the number of lost packets within a block of packets of fixed or variable size for several arrival models and several numbers of sessions. Numerical examples are provided to compare the distribution obtained with that obtained using the independence assumption to compute the loss probabilities of packets within a block. The results show that forward error correction schemes become less efficient due to the bursty nature of the packet loss processes; real-time traffic might be more sensitive to network congestion than was previously assumed; and the retransmission probability of ATM messages has been overestimated by the use of the independence assumption     

Routing strategies for maximizing throughput in LEO satellite networks

This paper develops routing and scheduling algorithms for packet transmission in a low Earth orbit satellite network with a limited number of transmitters and buffer space. We consider a packet switching satellite network, where time is slotted and the transmission time of each packet is fixed and equal to one time slot. Packets arrive at each satellite independently with a some probability during each time slot; their destination satellite is uniformly distributed. With a limited number of transmitters and buffer space on-board each satellite, contention for transmission inevitably occurs as multiple packets arrive at a satellite. First, we establish the stability region of the system in terms of the maximum admissible packet arrival rate that can possibly be supported. We then consider three transmission scheduling schemes for resolving these contentions: random packet win, where the winning packet is chosen at random; oldest packet win, where the packet that has traveled the longest distance wins the contention; and shortest hops win (SHW), where the packet closest to its destination wins the contention. We evaluate the performance of each of the schemes in terms of throughput. For a system without a buffer, the SHW scheme attains the highest throughput. However, when even limited buffer space is available, all three schemes achieve about the same throughput performance. Moreover, even with a buffer size of just a few packets the achieved throughput is close to that of the infinite buffer case.     

Performance comparison of different spread-spectrum signalingschemes for cellular mobile radio networks

Different spread-spectrum signaling schemes in a cellular mobile radio network are compared in terms of throughput and packet error probability. Bounds on the bit and packet error probabilities are derived for data modulation schemes with binary phase shift keying with noncoherent demodulation. Reed-Solomon coding is employed for error-correction purposes. In all cases, the effect of varying interference power (according to some inverse power of distance) of the desired signal, of the interfering signals, and of Rayleigh nonselective channel fading is accurately taken into account. The throughput in the mobile-to-base transmission mode is evaluated for the above data modulation, demodulation, and forward-error-control coding schemes. The comparison shows that, under the varying interference power model, the frequency-hopped scheme performs best among all schemes with the same bandwidth. Power control mechanisms are required to improve the performance of direct-sequence systems     

Cross-Layer Optimization of MAC and Network Coding in Wireless Queueing Tandem Networks

In wireless networks, throughput optimization is an essential performance objective that cannot be adequately characterized by a single criterion (such as the minimum transmitted or sum-delivered throughput) and should be specified over all source-destination pairs as a rate region. For a simple and yet fundamental model of tandem networks, a cross-layer optimization framework is formulated to derive the maximum throughput region for saturated multicast traffic. The contents of network flows are specified through network coding (or plain routing) in network layer and the throughput rates are jointly optimized in medium access control layer over fixed set of conflict-free transmission schedules (or optimized over transmission probabilities in random access). If the network model incorporates bursty sources and allows packet queues to empty, the objective is to specify the stability region as the set of maximum throughput rates that can be sustained with finite packet delay. Dynamic queue management strategies are used to expand the stability region toward the maximum throughput region. Network coding improves throughput rates over plain routing and achieves the largest gains for broadcast communication and intermediate network sizes. Throughput optimization imposes fundamental tradeoffs with transmission and processing energy costs such that the throughput-optimal operation is not necessarily energy efficient.     

A study of slot reuse in dual bus multiple access networks

In dual unidirectional bus networks, packets usually occupy fixed-length slots form the sending station to the end of the network. An erasure node is a specialized station which recognizes packets which have passed their destination stations and releases the slots for subsequent use. The authors derive the optimal locations for erasure nodes and show analytically, for uniform traffic, that only several erasure nodes are needed to achieve throughput close to twice the nominal network bandwidth. The results are tested by simulation of the DQDB (distributed queue dual bus) protocol, which demonstrates a realistic improvement of 40% with only three erasure nodes. Fair access among the stations is improved as well. The authors generalize the analytic results by providing an algorithm for determining the optimal erasure node locations and the throughput improvement, given any arbitrary traffic pattern. The application of this methodology to the related problem of bridged subnetworks is briefly discussed     

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.     

有线数字电视光分组交换网络物理层协议配置

从光分组的产生、编码方案、光分组的再生、光分组同步以及光分组交换的流量分配等入手,讨论了下一代有线数字电视光分组交换网络的物理结构、物理层协议配置及光-电、电-电的多种优化设计方案.     

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.