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.
The novelty of our work is the development of a computationally feasible network coding based retransmission algorithm whose gains are two-folded: 1) Our algorithm BENEFIT delivers better throughput with respect to the current best single-hop, NC based retransmission algorithm, and 2) we also demonstrate that our algorithm achieves minimum time to decode packets. None of the previous works @cite_3 - @cite_0 on NC based retransmission incorporates consideration of packet latency in their work. Here we will also show that it is no longer necessary to follow the packet coding rule @cite_4 , @cite_6 strictly. This relaxation in the coding rule has the potential for modification and development of other network coding based applications.
@cite_0 @cite_4 @cite_6 @cite_3
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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. 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. 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. 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.