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2008, 2008 5th IEEE International Conference on Mobile Ad Hoc and Sensor Systems
Network coding is a highly efficient data dissemination mechanism for wireless networks. Since network coded information can only be recovered after delivering a sufficient number of coded packets, the resulting decoding delay can become problematic for delay-sensitive applications such as real-time media streaming. Motivated by this observation, we consider several algorithms that minimize the decoding delay and analyze their performance by means of simulation. The algorithms differ both in the required information about the state of the neighbors' buffers and in the way this knowledge is used to decide which packets to combine through coding operations. Our results show that a greedy algorithm, whose encodings maximize the number of nodes at which a coded packet is immediately decodable significantly outperforms existing network coding protocols.
2009
We propose a linear network coding scheme to disseminate a finite number of data packets in arbitrary networks. The setup assumes a packet erasure channel, slotted time, and that nodes cannot transmit and receive information simultaneously. The dissemination process is completed when all terminals can decode the original data packets. We also assume a perfect knowledge of the information at each of the nodes, but not necessarily a perfect knowledge of the channel. A centralized controller decides which nodes should transmit, to what set of receiver nodes, and what information should be broadcasted. We show that the problem can be thought of as a scheduling problem, which is hard to solve. Thus, we consider the use of a greedy algorithm that only takes into account the current state of the system to make a decision. The proposed algorithm tries to maximize the impact on the network at each slot, i.e. maximize the number of nodes that will benefit from the coded packet sent by each active transmitter. We show that our scheme is considerably better, in terms of the number of slots to complete transmission, than schemes that choose the node with more information as the transmitter at every time slot.
Proceedings of the 13th annual ACM international conference on Mobile computing and networking - MobiCom '07, 2007
Network coding is seen as a promising technique to improve network throughput. In this paper, we study two important problems in localized network coding in wireless networks, which only requires each node to know about and coordinate with one-hop neighbors. In particular, we first establish a condition that is both necessary and sufficient for useful coding to be possible. We show this condition is much weaker than expected, and hence allows a variety of coding schemes to suit different network conditions and application preferences. Based on the understanding we establish, we are able to design a robust coding technique called loop coding that can improve network throughput and TCP throughput simultaneously.
2012
Abstract—To meet increasing throughput, delay and reliability demands, future wireless networks will have to rely on increased cooperation both among nodes and among protocol layers in a node. This requires a wide variety of knowledge, from theoretical issues such as graph theory or information theory, to issues at the physical and network layers. A potential solution is the recently proposed network coding because it offers a number of desired properties such as resource efficiency, robustness and security.
2014
Network coding is a technique that proposes a different approach for the protocol design in data communication networks. Thus, the nodes in the network are allowed not only to store and forward data packets, but also to process and mix different packets in a single coded packet. By using this technique, the throughput and robustness of the network can be significantly improved. However, the transmission delay of network coding is still not well understood. In real-time communication systems with stringent delay constraints, understanding the transmission delay distribution is at the core of implementing network coding in practical scenarios. Moreover, the benefits of network coding for broadcast scenarios have been proven, but the use of this technique in data gathering applications is limited. Unlike broadcast applications, where the main objective is to minimize the transmission delay, in data gathering applications the challenge is to reduce the data collection time, called the c...
IEEE Transactions on Communications, 2000
Intra-session network coding has been shown to offer significant gains in terms of achievable throughput and delay in settings where one source multicasts data to several clients. In this paper, we consider a more general scenario where multiple sources transmit data to sets of clients and study the benefits of inter-session network coding, when network nodes have the opportunity to combine packets from different sources. In particular, we propose a novel framework for optimal rate allocation in inter-session network coding systems. We formulate the problem as the minimization of the average decoding delay in the client population and solve it with a gradient-based stochastic algorithm. Our optimized inter-session network coding solution is evaluated in different network topologies and compared with basic intra-session network coding solutions. Our results show the benefits of proper coding decisions and effective rate allocation for lowering the decoding delay when the network is used by concurrent multicast sessions.
EURASIP Journal on Wireless Communications and Networking, 2017
In traditional store and forward protocols, lost packets have no impact on the delivery of other transmitted packets. With network coding, the impact of a packet loss may affect the decoding of other transmitted packets thus affecting the entire process of communication between nodes. In this work, we propose a new network coding model that allows generating, coding, decoding and transmission activities on the packets. Based on this model, the impact of lost packets on buffering and the complexity at the receiving nodes is studied and two new mechanisms are proposed to allow the recovery of lost packets. Compared to traditional linear network coding protocol, our mechanisms provide a significant performance amelioration in terms of number of transmissions required to recover from packet loss.
In recent years, network coding has become one of the most interesting fields and has attracted considerable attention from both industry and academia. The idea of network coding is based on the concept of allowing intermediate nodes to encode and combine incoming packets instead of only copy and forward them. This approach, by augmenting the multicast and broadcast efficiency of multi-hop wireless networks, increases the capacity of the network and improves its throughput and robustness. While a wide variety of papers described applications of network coding in different types of networks such as delay tolerant networks, peer to peer networks and wireless sensor networks, the detailed practical implementation of network coding has not been noted in most papers. Since applying network coding in real scenarios requires an acceptable understanding of mathematics and algebra, especially linear equations, reduced row echelon matrices, field and its operations, this paper provides a comprehensive guidance for the implementation of almost all required concepts in network coding. The paper explains the implementation details of network coding in real scenarios and describes the effect of the field size on network coding.
MILCOM 2007 - IEEE Military Communications Conference, 2007
This paper investigates the interaction between network coding and link-layer transmission rate diversity in multihop wireless networks. By appropriately mixing data packets at intermediate nodes, network coding allows a single multicast flow to achieve higher throughput to a set of receivers. Broadcast applications can also exploit link-layer rate diversity, whereby individual nodes can transmit at faster rates at the expense of corresponding smaller coverage area. We first demonstrate how combining rate-diversity with network coding can provide a larger capacity for data dissemination of a single multicast flow, and how consideration of rate diversity is critical for maximizing system throughput. We also study the impact of both network coding and rate diversity on the dissemination latency for a class of quasi real-time applications, where the freshness of disseminated data is important. Our results provide evidence that network coding may lead to a latency-vs-throughput tradeoff in wireless environments, and that it is thus necessary to adapt the degree of network coding to ensure conformance to both throughput and latency objectives. There is an increasing interest in understanding the potential performance gains accruing from the use of network coding in multi-hop wireless environments. In particular, many military battlefield scenarios exhibit two characteristics that appear to motivate the use of network coding: a) the reliance on bandwidth-constrained, ad-hoc wireless links (e.g. using MANETs formed by vehicle-mounted radios in urban insurgencies) and b) the need to disseminate information (e.g., maps, mission commands) to multiple recipients. The initial results on the power of network coding NC, such as the original demonstration in [1] of how in-network mixing of packets by intermediate nodes helps to achieve a communication capacity that is not achievable solely through routing, were obtained for the case of a lossless, wireline network. More recently, several groups have investigated the potential performance gains realized by network coding for both
2013
Network coding has shown a good potential for improving network throughput and, recently, has been used in multimedia transmission over wireless networks. In addition to guaranteed throughput, real-time multimedia transmission requires bounded delay, delay variation, and packet loss. This paper presents RTOC, a new XOR-based opportunistic network coding architecture for real-time data transmission. RTOC is an application-independent architecture that takes into consideration the characteristics of real-time traffic and provides an efficient framework to optimize multimedia application requirements such as bandwidth, delay, delay variation, and loss. The performance of RTOC was evaluated using simulated video streaming traffic. The results demonstrated that RTOC can improve the real-time performance metrics such as the packet delivery ratio, end-to-end delay, jitter, and throughput by considerable margins.
2012
In wireless mesh networks for unicast traffic, opportunistic network coding are specifically introduced for improved network utilization. Some recent practical implementations like COPE, BEND, DCAR and DODE have shown the promising results over the original 802.11 conventional forwarding mechanism. For better performance, the aim of all mentioned network coding implementations is trying to find more and more coding chances in network topologies. However, they restricted the finding within a simple rule "a pair of coder and decoder". For every coded packet (the combination of natives packets of traffic flows) sent on the traffic flow, we always have one coder (which creates the coded packet) and one decoder (which retrieves the desired packet for the destination). The trivial rule limited coding chances much. In this paper, we are loosing this noose: we decouple the coding and decoding functions from strictly a pair of coder and decoder. For one coder, we can have multiple decoders on the path (up to the number of other traffic flows involved in the coding). With this, more coding chances are found, thus, improving the network performance. We extend our proposed DODE with this new idea, called Distributed Opportunistic and Diffused Coding with Multiple Decoders-DODEX. We implement DODEX system in NS-2. The simulation results show that DODEX can outperform its previous introducing systems.
IEEE/ACM Transactions on Networking, 2013
Network coding has been proposed as a technique that can potentially increase the transport capacity of a wireless network via mixing data packets at intermediate routers. However, most previous studies either assume a fixed transmission rate or do not consider the impact of using diverse rates on the network coding gain. Since in many cases, network coding implicitly relies on overhearing, the choice of the transmission rate has a big impact on the achievable gains. The use of higher rates works in favor of increasing the native throughput. However, it may in many cases work against effective overhearing. In other words, there is a tension between the achievable network coding gain and the inherent rate gain possible on a link. In this paper, our goal is to drive the network toward achieving the best tradeoff between these two contradictory effects. We design a distributed framework that: 1) facilitates the choice of the best rate on each link while considering the need for overhearing; and 2) dictates the choice of which decoding recipient will acknowledge the reception of an encoded packet. We demonstrate that both of these features contribute significantly toward gains in throughput. We extensively simulate our framework in a variety of topological settings. We also fully implement it on real hardware and demonstrate its applicability and performance gains via proof-of-concept experiments on our wireless testbed. We show that our framework yields throughput gains of up to 390% as compared to what is achieved in a rate-unaware network coding framework.
IEEE Journal on Selected Areas in Communications, 2009
This paper focuses on using Network Coding (NC) with TCP for multi-hop wireless networks to provide faulttolerant and timely delivery of streaming data. The paper shows that there is an inherent latency in video playback when TCP with random linear NC is employed. With the objective of reducing latency and jitter at the receiver, the paper proposes a Variable Bucket size based Network Coding (VBNC) technique that modifies the TCP congestion control to adapt to the arriving traffic and dynamic network conditions. Simulation results demonstrate that on an average the proposed algorithm reduces observed latency at playback by 80% and jitter by more than 50% over standard TCP. More importantly, a significant reduction in the initial start-up delay is observed which enhances the performance of streaming services.
—In this paper, we introduce network coding (NC) at intermediate nodes of a linear network where coding is conducted on address correlated packets. In particular, we propose a novel algorithm for decoding network-coded messages that enhances the bandwidth usage and reduces the resource allocation by allowing nodes in the network to perform distributed decoding. We show that, compared to traditional decoding at end nodes, distributed decoding allows reduction not only in the number of transmitted bytes but also in the number of resources needed to perform NC. Our strategy is compared to decoding at end nodes and experiments show a reduction of 64% in the number of transmitted bytes and a reduction of 93% of buffering time when using distributed decoding to exchange 1000 packets between end nodes of a linear network of 8 nodes.
The paper focuses on using Network Coding (NC) with TCP for multi-hop wireless networks to provide fault-tolerant and timely delivery of streaming data. The paper shows that there is an inherent latency in video playback when TCP with random linear NC is employed. With the objective of reducing latency and jitter at the receiver, the paper proposes a Variable Bucket size based Network Coding (VBNC) technique that modifies the TCP congestion control to adapt to the arriving traffic and dynamic network conditions. Simulation results demonstrate that on an average the proposed algorithm reduces observed latency at playback by 80% and jitter by more than 50% over standard TCP. More importantly, a significant reduction in the initial start-up delay is observed which enhances the performance of streaming services.
Wireless Personal Communications, 2019
This paper proposed a novel algorithm called Efficiency Network Coding (ENC) for wireless mesh networks. The ENC algorithm is based on COPE protocol and consists of two parts: In the first part, ENC codes the packets which have a smaller difference in size than other packets and so uses less bandwidth to code and send packets. In the second part, the nodes which their packets are in front of virtual queue and the ones which their packets are in front of output queue participate in coding. Therefore, ENC increases coding opportunities and takes less time for sending packets. The proposed ENC is implemented in NS2.34 with TCP-New Reno on COPE framework and compared with the throughput of TCP in COPE. The results of this paper show that the throughput of the proposed ENC is greater than that of COPE in most cases. The time complexity of ENC depends on the overhear packets in the nodes and can be less, equal, or more than COPE in different conditions.
IEEE Transactions on Vehicular Technology, 2017
In recent years, network coding has emerged as an innovative method that helps a wireless network approach its maximum capacity, by combining multiple unicasts in one broadcast. However, the majority of research conducted in this area is yet to fully utilize the broadcasting nature of wireless networks, and still assumes fixed route between the source and destination that every packet should travel through. This assumption not only limits coding opportunities, but can also cause buffer overflow in some specific intermediate nodes. Although some studies considered scattering of the flows dynamically in the network, they still face some limitations. This paper explains pros and cons of some prominent research in network coding and proposes a Flexible and Opportunistic Network Coding scheme (FlexONC) as a solution to such issues. Furthermore, this research discovers that the conditions used in previous studies to combine packets of different flows are overly optimistic and would affect the network performance adversarially. Therefore, we provide a more accurate set of rules for packet encoding. The experimental results show that FlexONC outperforms previous methods especially in networks with high bit error rate, by better utilizing redundant packets spread in the network.
2015
Over the past decade, network coding (NC) has emerged as a new paradigm for data communications and has attracted much popularity and research interest in information and coding theory, networking, wireless communications and data storage. Random linear NC (RLNC) is a subclass of NC that has shown to be suitable for a wide range of applications thanks to its desirable properties, namely throughput-optimality, simple encoder design and efficient operation with minimum feedback requirements. However, for delay-sensitive applications, the mentioned advantages come with two main issues that may restrict RLNC usage in practice. First is the trade-off between the delay and throughput performances of RLNC, which can adversely affect the throughput-optimality of RLNC and hence the overall performance of RLNC. Second is the usage of feedback, where even if feedback is kept at minimum it can still incur large amount of delay and thus degrade the RLNC performance, if not optimized properly. In...
IEEE INFOCOM 2009 - The 28th Conference on Computer Communications, 2009
Motivated by streaming applications with stringent delay constraints, we consider the design of online network coding algorithms with timely delivery guarantees. Assuming that the sender is providing the same data to multiple receivers over independent packet erasure channels, we focus on the case of perfect feedback and heterogeneous erasure probabilities. Based on a general analytical framework for evaluating the decoding delay, we show that existing ARQ schemes fail to ensure that receivers with weak channels are able to recover from packet losses within reasonable time. To overcome this problem, we redefine the encoding rules in order to break the chains of linear combinations that cannot be decoded after one of the packets is lost. Our results show that sending uncoded packets at key times ensures that all the receivers are able to meet specific delay requirements with very high probability.
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