Retransmission based on packet acknowledgement (ACK NAK) is a fundamental error control technique employed in IEEE 802.11–2007 unicast network. However the 802.11–2007 standard falls short of proposing a reliable MAC-level recovery protocol for multicast frames. In this paper we propose a latency and bandwidth efficient coding algorithm based on the principles of network coding for retransmitting lost packets in a singlehop wireless multicast network and demonstrate its effectiveness over previously proposed network coding based retransmission algorithms.
Packet retransmission based on network coding for a one-to-many, single-hop multicast network is a recent field of study, first proposed by D. @cite_1 , which was later further elaborated into @cite_7 by D. @cite_7 the authors demonstrate bandwidth effectiveness achieved by employing greedy network coding for retransmission over traditional ARQ schemes through simulation work. @cite_3 the authors follow up the work in @cite_1 by comparing various packet coding algorithms for packet retransmissions. While in @cite_8 , the authors presents an analytical work on the reliability performance of network coding compared with ARQ and FEC in a lossy network. Network Coded Piggy Back (NCPB) @cite_2 demonstrates an efficient and practical testbed implemented random linear network coding based many-to-many reliable network model for real-time multi-player game network. Since our work primarily focuses on proposing an efficient network coding based retransmission algorithm for a one-to-many single-hop network, we will be comparing our results with the algorithm given in @cite_3 which is the most closely related work.
@cite_7 @cite_8 @cite_1 @cite_3 @cite_2
2119011159 2130286537 2165683708 2150088023
The capacity gain of network coding has been extensively studied in wired and wireless networks. Recently, it has been shown that network coding improves network reliability by reducing the number of packet retransmissions in lossy networks. However, the extent of the reliability benefit of network coding is not known. This paper quantifies the reliability gain of network coding for reliable multicasting in wireless networks, where network coding is most promising. We define the expected number of transmissions per packet as the performance metric for reliability and derive analytical expressions characterizing the performance of network coding. We also analyze the performance of reliability mechanisms based on rateless codes and automatic repeat request (ARQ), and compare them with network coding. We first study network coding performance in an access point model, where an access point broadcasts packets to a group of K receivers over lossy wireless channels. We show that the expected number of transmissions using ARQ, compared to network coding, scales as ominus (log K) as the number of receivers becomes large. We then use the access point model as a building block to study reliable multicast in a tree topology. In addition to scaling results, we derive expressions for the expected number of transmissions for finite multicast groups as well. Our results show that network coding significantly reduces the number of retransmissions in lossy networks compared to an ARQ scheme. However, rateless coding achieves asymptotic performance results similar to that of network coding. Traditional approaches to reliably transmit information over an error-prone network employ either forward error correction (FEC) or retransmission techniques. In this paper, we propose some network coding schemes to reduce the number of broadcast transmissions from one sender to multiple receivers. The main idea is to allow the sender to combine and retransmit the lost packets in a certain way so that with one transmission, multiple receivers are able to recover their own lost packets. For comparison, we derive a few theoretical results on the bandwidth efficiency of the proposed network coding and traditional automatic repeat-request (ARQ) schemes. Both simulations and theoretical analysis confirm the advantages of the proposed network coding schemes over the ARQ ones. This paper investigates Automatic Repeat request (ARQ) designs for Physical-layer Network Coding (PNC) systems. Most prior work related to PNC explores its use in Two-Way Relay Channel (TWRC). We have previously found that, besides TWRC, there are many other PNC building blocks—building blocks are simple small network structures that can be used to construct a large network. In some of these PNC building blocks, the receivers can obtain side information through overhearing. Although such overheard information is not the target information that the receivers desire, the receivers can exploit the overheard information together with a network-coded packet received to obtain a desired native packet. This can yield substantial throughput gain. Our previous study, however, assumed what is sent always gets received. In practice, that is not the case. Error control is needed to ensure reliable communication. This paper focuses on ARQ designs for ensuring reliable PNC communication. The availability of overheard Information and its potential exploitation make the ARQ design of a network-coded system different from that of a non-network-coded system. In this paper, we lay out the fundamental considerations for such ARQ designs: 1) we put forth a framework to track the stored coded packets and overheard packets to increase the chance of packet extraction, and derive the throughput gain achieved therefore; 2) we investigate two variations of PNC ARQ, coupled and non-coupled ARQs, and prove that non-coupled ARQ is more efficient; 3) we show how to optimize parameters in PNC ARQ—specifically the window size and the ACK frequency—to minimize the throughput degradation caused by ACK feedback overhead and wasteful retransmissions due to lost ACK. In a document network such as a citation network of scientific documents, web-logs, etc., the content produced by authors exhibits their interest in certain topics. In addition some authors influence other authors' interests. In this work, we propose to model the influence of cited authors along with the interests of citing authors. Moreover, we hypothesize that apart from the citations present in documents, the context surrounding the citation mention provides extra topical information about the cited authors. However, associating terms in the context to the cited authors remains an open problem. We propose novel document generation schemes that incorporate the context while simultaneously modeling the interests of citing authors and influence of the cited authors. Our experiments show significant improvements over baseline models for various evaluation criteria such as link prediction between document and cited author, and quantitatively explaining unseen text. The severe nature of underwater channel poses a great challenge for prolonging underwater acoustic sensor networks (UASNs) lifetime and achieving a reliable communication performance. Traditional approaches to improve the reliability such as automatic repeat request (ARQ) negatively affect the network lifetime (NL) due to energy dissipation caused by ARQ retransmission. A forward error correction (FEC) technique called fountain codes (FCs) can solve the energy efficiency problem of ARQ by transmitting both the original packet and some redundant packets to ensure a targeted reliability with few or no retransmissions. In this paper, we investigate performances of both traditional ARQ- and FC-based FEC methods in terms of NL, end-to-end delay, energy consumption, and frame error rate (FER) for UASNs. In this context, we abstract energy dissipation characteristics of conventional ARQ- and FC-based FEC method at the link-layer. We propose an integer linear programming (ILP) framework that maximizes the NL, which is operated on top of developed link-layer energy consumption models. Our results reveal that FC-based FEC methods can prolong the NL at a minimum of 16 while end-to-end delay, energy consumption, and FER can be reduced at least by 11 , 14 , and 9 as compared to classical ARQ, respectively. A multi-player video game via wireless connections using portable devices is one of the most popular applications of ad-hoc networks. Broadcast transmissions can be used for many-to-many communications in multi-player games. However, reliable broadcast communications are hard to realize because of the lack of retransmissions in the medium access control (MAC) layer. In this paper, we propose a broadcast method using random linear network coding in order to enhance the reliability of many-to-many and real-time communications. Exploiting the periodic nature of game traffic as well as inherent robustness offered by random network coding, the proposed method can provide reliable packet deliveries between 2 nodes which have a link under constant fade or even under temporary loss. Our simulation results show that the proposed method can provide higher reliability than the other schemes using multi point relay (MPR) or redundant transmissions such as forward error correction (FEC). We also implement the proposed method in a wireless testbed, and show that the proposed method achieves high reliability in a real-world environment with practical degree of complexity when installed on current wireless devices. Based on the changeability of wireless network circumstance, information packets are prone to lose in the process of the wireless transmission. In wireless broadcasting ,any node of multi-nodes request retransmission information packets .This paper presents a novel retransmission approach in wireless broadcasting based on network coding (NCWBR), whose key idea is to combine different lost packets with network coding to achieve retransmission. Theoretical analysis reveals that our approach can ensure the solvability in the received nodes, and have better retransmission performance. Simulation results indicate that compared with existing approach, the approach in this paper can effectively reduce the average number of transmissions and advance the transmission efficiency. Delay-reliability (D-R), and throughput-delay-reliability (T-D-R) tradeoffs in an ad hoc network are derived for single hop and multi-hop transmission with automatic repeat request (ARQ) on each hop. The delay constraint is modeled by assuming that each packet is allowed at most D retransmissions end-to-end, and the reliability is defined as the probability that the packet is successfully decoded in at most D retransmissions. The throughput of the ad hoc network is characterized by the transmission capacity, where the transmission capacity is defined to be the maximum density of spatial transmissions that can be simultaneously supported in an ad hoc network under quality of service (QoS) constraints (maximum retransmissions and reliability). The transmission capacity captures the T-D-R tradeoff as it incorporates the dependence between the throughput, the maximum delay, and the reliability. Given an end-to-end retransmission constraint of D, the optimal allocation of the number of retransmissions allowed at each hop is derived that maximizes a lower bound on the transmission capacity. Optimizing over the number of hops, single hop transmission is shown to be optimal for maximizing a lower bound on the transmission capacity in the sparse network regime. To address the problem of information spreading over lossy communication channels, this paper proposes a joint FoUntain coding and Network coding (FUN) approach. Different from the Transmission Control Protocol (TCP), our FUN approach is a mechanism of Forward Error Correction (FEC), which does not use retransmission for recovery of lost packets. The novelty of our FUN approach lies in combining the best features of fountain coding, intra-session network coding, and cross-next-hop network coding. As such, our FUN approach is capable of achieving unprecedented high throughput over lossy channels. Experimental results demonstrate that our FUN approach achieves higher throughput than the existing schemes for multihop wireless networks. The theory of network coding promises significant benefits in network performance, especially in lossy networks and in multicast and multipath scenarios. To realize these benefits in practice, we need to understand how coding across packets interacts with the acknowledgment (ACK)-based flow control mechanism that forms a central part of today's Internet protocols such as transmission control protocol (TCP). Current approaches such as rateless codes and batch-based coding are not compatible with TCP's retransmission and sliding-window mechanisms. In this paper, we propose a new mechanism called TCP NC that incorporates network coding into TCP with only minor changes to the protocol stack, thereby allowing incremental deployment. In our scheme, the source transmits random linear combinations of packets currently in the congestion window. At the heart of our scheme is a new interpretation of ACKs-the sink acknowledges every degree of freedom (i.e., a linear combination that reveals one unit of new information) even if it does not reveal an original packet immediately. Thus, our new TCP ACK rule takes into account the network coding operations in the lower layer and enables a TCP-compatible sliding-window approach to network coding. Coding essentially masks losses from the congestion control algorithm and allows TCP NC to react smoothly to losses, resulting in a novel and effective approach for congestion control over lossy networks such as wireless networks. An important feature of our solution is that it allows intermediate nodes to perform re-encoding of packets, which is known to provide significant throughput gains in lossy networks and multicast scenarios. Simulations show that our scheme, with or without re-encoding inside the network, achieves much higher throughput compared to TCP over lossy wireless links. We present a real-world implementation of this protocol that addresses the practical aspects of incorporating network coding and decoding with TCP's window management mechanism. We work with TCP-Reno, which is a widespread and practical variant of TCP. Our implementation significantly advances the goal of designing a deployable, general, TCP-compatible protocol that provides the benefits of network coding. Wireless LANs (WLANs) have been deployed at a remarkable rate at university campuses, office buildings, airports, hotels, and malls. Providing efficient and reliable wireless communications is challenging due to inherent lossy wireless medium and imperfect packet scheduling that results in packet collisions. In this paper, we develop an efficient retransmission scheme (ER) for wirless LANs. Instead of retransmitting the lost packets in their original forms, ER codes packets lost at different destinations and uses a single retransmission to potentially recover multiple packet losses. We develop a simple and practical protocol to realize the idea and implement it in both simulation and testbed, and our results demonstrate the effectiveness of this approach. Mobile Communications DepartmentEur´ecom InstituteSophia-Antipolis, Franceraymond.knopp@eurecom.frAbstractIn this work, we are interested in characterizing the performance of decentral-ized multiple-access and retransmission schemes for multi-hop wireless networks.We assume that nodes are randomly distributed on the plane according to a homo-geneous spatial Poisson process. We investigate two transmission strategies: onethat maximizes the distance of a successful transmission for each hop towards theﬁnal destination; and the other that forwards packets to the closest node in range inthe direction of the ﬁnal destination where transmission is successful. This is usefulin the context of multi-hop wireless networks as it is important to assess the trade-oﬀ between spatial density of communication and the range of each transmission.The results of this work also show that coding and retransmissions provide meansof reliable communication coupled with a completely decentralized multiple-accessstrategy. In this work we consider coding over packets that randomly arrive to a source node for transmission to a single destination. We present a queueing model for a random linear coding scheme that adapts to the amount of traffic at the source node. If there is only one packet in the queue when the channel becomes free, then reliable transmission is carried out by retransmitting lost packets. If there are at least two packets in the queue, then random linear coding is carried out over the number of packets available in the queue when the channel becomes free. We provide a bulk-service queuing model and results on the delay of this random linear coding scheme, and show that its delay performance asymptotically approaches that of a retransmission scheme. This paper presents the evaluation of the multi-hop aggregate information efficiency of the slotted and unslotted ALOHA protocols. We consider a multi-hop wireless network where the nodes are spatially characterized by a Poisson point process and the traffic generation also follows a Poisson distribution. By applying the properties of stochastic geometry, we derive a closed-form lower bound on the outage probability as a function of the required communication rate, the single-hop distance, the number of hops and the maximum number of retransmissions. The results indicate that slotted ALOHA always outperforms its unslotted version, demonstrating the importance of synchronization in distributed networks. In addition, we show that it is always possible to optimize the network efficiency by properly setting the required rate for a given packet density. Finally, in the scenario considered, the use of retransmissions and multiple hops never achieves the best performance if compared to the option of single-hop links without retransmissions. In mobile packet data and voice networks, a special coding scheme, known as the incremental redundancy hybrid ARQ (IR HARQ), achieves higher throughput efficiency than ordinary turbo codes by adapting its error correcting code redundancy to fluctuating channel conditions characteristic for this application. An IR HARQ protocol operates as follows. Initially, the information bits are encoded by a “mother” code and a selected number of parity bits are transmitted. If a retransmission is requested, only additional selected parity bits are transmitted. At the receiving end, the additional parity bits are combined with the previously received bits, allowing for an increase in the error correction capacity. This procedure is repeated after each subsequent retransmission request until the entire codeword of the mother code is transmitted. A number of important issues such as error rate performance after each transmission on time varying channels, and rate and power control are difficult to analyze in a network employing a particular HARQ scheme, i.e., a given mother code and given selection of bits for each transmission. By relaxing only the latter constraint, namely, by allowing random selection of the bits for each transmission, we provide very good estimates of error rates allowing us to address to a certain extent the rate and power control problem. Current approaches to the practical implementation of network coding are batch-based, and often do not use feedback, except possibly to signal completion of a file download. In this paper, the various benefits of using feedback in a network coded system are studied. It is shown that network coding can be performed in a completely online manner, without the need for batches or generations, and that such online operation does not affect the throughput. Although these ideas are presented in a single-hop packet erasure broadcast setting, they naturally extend to more general lossy networks which employ network coding in the presence of feedback. The impact of feedback on queue size at the sender and decoding delay at the receivers is studied. Strategies for adaptive coding based on feedback are presented, with the goal of minimizing the queue size and delay. The asymptotic behavior of these metrics is characterized, in the limit of the traffic load approaching capacity. Different notions of decoding delay are considered, including an order-sensitive notion which assumes that packets are useful only when delivered in order. Our work may be viewed as a natural extension of Automatic Repeat reQuest (ARQ) schemes to coded networks. In this paper, the fundamental performance tradeoff of the delay-limited multiple-input multiple-output (MIMO) automatic retransmission request (ARQ) channel is explored. In particular, we extend the diversity-multiplexing tradeoff investigated by Zheng and Tse in standard delay-limited MIMO channels with coherent detection to the ARQ scenario. We establish the three-dimensional tradeoff between reliability (i.e., diversity), throughput (i.e., multiplexing gain), and delay (i.e., maximum number of retransmissions). This tradeoff quantifies the ARQ diversity gain obtained by leveraging the retransmission delay to enhance the reliability for a given multiplexing gain. Interestingly, ARQ diversity appears even in long-term static channels where all the retransmissions take place in the same channel state. Furthermore, by relaxing the input power constraint allowing variable power levels in different retransmissions, we show that power control can be used to dramatically increase the diversity advantage. Our analysis reveals some important insights on the benefits of ARQ in slow-fading MIMO channels. In particular, we show that 1) allowing for a sufficiently large retransmission delay results in an almost flat diversity-multiplexing tradeoff, and hence, renders operating at high multiplexing gain more advantageous; 2) MIMO ARQ channels quickly approach the ergodic limit when power control is employed. Finally, we complement our information-theoretic analysis with an incremental redundancy lattice space-time (IR-LAST) coding scheme which is shown, through a random coding argument, to achieve the optimal tradeoff(s). An integral component of the optimal IR-LAST coding scheme is a list decoder, based on the minimum mean-square error (MMSE) lattice decoding principle, for joint error detection and correction. Throughout the paper, our theoretical claims are validated by numerical results In this work we address the stability and delay performance of a multicast erasure channel with random arrivals at the source node. We consider both a standard retransmission (ARQ) scheme as well as random linear coding. Our results show that while random linear coding may outperform re-transmissions for heavy traffic, the delay incurred by the use of random linear codes is significantly higher when the source is lightly loaded The future wireless networks envision ultra-reliable communication with efficient use of the limited wireless channel resources. Closed-loop repetition protocols where retransmission of a packet is enabled using a feedback channel has been adopted since early days of wireless telecommunication. Protocols such as automatic repeat request (ARQ) are used in today's wireless technologies as a mean to provide the link with reduced rate of packet outage and increased average throughput. The performance of such protocols is strongly dependent to the feedback channel reliability. This paper studies the problem of feedback error and proposes a new method of acknowledging packet delivery for retransmission protocols in unreliable feedback channel conditions. The proposed method is based on backwards composite acknowledgment from multiple packets in a retransmission protocol and provides the scheduler of the wireless channel with additional parameters to configure ultra-reliable communication for a user depending on channel quality. Numerical analysis are presented which show orders of magnitude increase in reliability of the proposed method as compared to ARQ at the cost of a small increase in average experienced delay. Scholars are increasingly citing electronic “web references” which are not preserved in libraries or full text archives. WebCite is a new standard for citing web references. To “webcite” a document involves archiving the cited Web page through www.webcitation.org and citing the WebCite permalink instead of (or in addition to) the unstable live Web page. This journal has amended its “instructions for authors” accordingly, asking authors to archive cited Web pages before submitting a manuscript. Almost 200 other journals are already using the system. We discuss the rationale for WebCite, its technology, and how scholars, editors, and publishers can benefit from the service. Citing scholars initiate an archiving process of all cited Web references, ideally before they submit a manuscript. Authors of online documents and websites which are expected to be cited by others can ensure that their work is permanently available by creating an archived copy using WebCite and providing the citation information including the WebCite link on their Web document(s). Editors should ask their authors to cache all cited Web addresses (Uniform Resource Locators, or URLs) “prospectively” before submitting their manuscripts to their journal. Editors and publishers should also instruct their copyeditors to cache cited Web material if the author has not done so already. Finally, WebCite can process publisher submitted “citing articles” (submitted for example as eXtensible Markup Language [XML] documents) to automatically archive all cited Web pages shortly before or on publication. Finally, WebCite can act as a focussed crawler, caching retrospectively references of already published articles. Copyright issues are addressed by honouring respective Internet standards (robot exclusion files, no-cache and no-archive tags). Long-term preservation is ensured by agreements with libraries and digital preservation organizations. The resulting WebCite Index may also have applications for research assessment exercises, being able to measure the impact of Web services and published Web documents through access and Web citation metrics. @PARASPLIT [J Med Internet Res 2005;7(5):e60]