Reliable data transmission is desirable in wireless sensor networks due to the high packet loss rate during multi-hop transmissions. To reliably transmit data for event-driven applications, packet loss recovery mechanism is needed. For loss recovery, sensor nodes need to keep packets in their buffers until transmissions successfully complete. However, since sensor nodes have limited memory, packets cannot be buffered for a long period of time. This letter proposes an efficient buffer management technique that caches data packets for appropriate amount of time to minimize the resource requirements and at the same time provide reliable data transmission among sensor nodes.

Traditional TCP has a good congestion control strategy that adapts its sending rate in accordance with network congestion. In addition, a fast recovery algorithm can help TCP achieve better throughput by responding to temporary network congestion well. However, if multiple packet losses occur, the time to enter congestion avoidance phase would be delayed due to the long recovery time. Moreover, during the recovery phase, TCP freezes congestion window size until all lost packets are recovered, and this can make recovery time much longer resulting in performance degradation. To mitigate such recovery overhead, we propose Momentary recovery algorithm that recovers packet loss without an extra recovery phase. As other TCP and variants, our algorithm also halves the congestion window size when packet drop is detected, but it performs congestion avoidance phase immediately as if loss recovery is completed. For lost packets, TCP sender transmits them along with normal packets as long as congestion window permits rather than performs fast retransmission. In this manner, we can eliminate recovery overhead efficiently and reach steady state momentarily after network congestion. Finally, we provide a simulation based study on TCP recovery behaviors and confirm that our Momentary recovery algorithm always shows better performance compared with NewReno, SACK, and FACK.

This letter proposes a new retransmission persistence management scheme for selective repeat automatic repeat request (SR-ARQ). By considering the overall traffic load that has to be managed by SR-ARQ, the proposed scheme arbitrates the retransmission persistence to prevent an abrupt delay increment due to excessive link-level local retransmissions. OPNET simulations show that SR-ARQ performs better with the proposed scheme than with a fixed value of retransmission persistence in terms of the throughput of transmission control protocol (TCP).

In this paper, we study the end-to-end TCP performance over a path deploying a High-Availability cluster, whose characteristics are highlighted by the failover procedure to remove single-point failure. This paper proposes an approach, called High-Availability Local Recovery (HALR), to enhance TCP performance in the face of a cluster failover. To minimize the latency of retransmission, HALR saves TCP packets selectively and resends them locally after the failover is finished. For better understanding, we further develop simple analytic models to predict the TCP performance in the aspect of flow latency under a range of failover times and the effects of HALR. Using simulation results, we validate our models and show that HALR improves the TCP performance significantly over a failover event as compared with the original TCP. Typically, HALR reduces the flow latency from 4.1 sec to less than 1.9 sec when the failover time equals to 500 ms. The simulation by real packet trace further demonstrates that the memory requirement of the proposed solution is not a concern for modern network equipments.

In this paper, we consider consecutive burst transmission with burst loss recovery based on Forward Error Correction (FEC) in which redundant data is transmitted with multiple bursts. We propose two burst generation methods: Out-of Burst Generation (OBG) and In-Burst Generation (IBG). The OBG generates a redundant burst from redundant data, while the IBG reconstructs a burst from an original data block and a part of the redundant data. For both methods, the resulting bursts are transmitted consecutively. If some bursts among the bursts are lost at an intermediate node, the lost bursts can be recovered with the redundant data using FEC processing at the destination node. We evaluate by simulation the proposed methods in a uni-directional ring network and NSFNET, and compare the performances of the proposed methods with the extra-offset time method. Numerical examples show that the proposed methods can provide a more reliable transmission than the extra-offset time method for the OBS network where the maximum number of hops is large. Moreover, it is shown that the end-to-end transmission delay for our proposed methods can be decreased by enhancing the FEC processor or by increasing the number of FEC processors.

It has been a very important issue to evaluate the performance of transmission control protocol (TCP), and the importance is still growing up because TCP will be deployed more widely in future wireless as well as wireline networks. It is also the reason why there have been a lot of efforts to analyze TCP performance more accurately. Most of these works are focusing on overall TCP end-to-end throughput that is defined as the number of bytes transmitted for a given time period. Even though each TCP's fast recovery strategy should be considered in computation of the exact time period, it has not been considered sufficiently in the existing models. That is, for more detailed performance analysis of a TCP implementation, the fast recovery latency during which lost packets are retransmitted should be considered with its relevant strategy. In this paper, we extend the existing models in order to capture TCP's loss recovery behaviors in detail. On the basis of the model, the loss recovery latency of three TCP implementations can be derived with considering the number of retransmitted packets. In particular, the proposed model differentiates the loss recovery performance of TCP using selective acknowledgement (SACK) option from TCP NewReno. We also verify that the proposed model reflects the precise latency of each TCP's loss recovery by simulations.

In packet networks including the Internet and commercial 3G wireless bearers, the network states that a streaming media application experiences are not known a priori and exhibit time-varying characteristics. For such dynamic environments, network-adaptive techniques are essential to efficiently deliver video data. In this paper, we propose a frame-based optimal scheduling algorithm which incorporates a MAP (Maximum A Posteriori) framework for adapting to varying network loss rate. The optimal transmission schedule is determined such that effective frame-rate is maximized at playback. Also, for multiple packets per frame, frame-based selection of delivery order greatly reduces computational complexity for a server scheduler when compared with packet-based scheduling techniques. In addition, by dynamically estimating instantaneous packet loss probability, the proposed scheduler performs network-adaptive transmission for streaming video over time-varying packet networks. Simulation results for test video sequence show that the proposed scheduling algorithm outperforms conventional ARQ-based schemes from a view point of reconstructed video quality as well as playable frame-rate. In particular, the proposed scheduling algorithm exhibits significant improvements of frame-rate over highly lossy channels.

This paper proposes a packet loss recovery method using packets arrived behind the playout time for CELP (Code Excited Liner Prediction) decoding. The proposed method recovers synchronization of the filter states between encoding and decoding in the period following packet loss. The recovery is performed by replacing the degraded filter states with the ones calculated from the late arrival packet in decoding. When the proposed method is applied to the AMR (Adaptive Multi-Rate) speech decoder, it improves the segmental SNR (Signal-to-Noise Ratio) by 0.2 to 1.8 dB at packet loss rates of 2 to 10 % in case that all the packet losses occur due to their late arrival. PESQ (Perceptual Evaluation of Speech Quality) results also show that the proposed method slightly improves the speech quality. The subjective test results show that five-grade mean opinion scores are improved by 0.35 and 0.28 at a packet loss rate of 5 % at speech coding bitrates of 7.95 and 12.2 kbit/s, respectively.

A new algorithm is presented for recovering the blocks lost with the cell loss in the ATM transmission of the images coded by Jacquin's fractal coding. The key technique of the proposed BLRA (block loss recovery algorithm) is a fractal extrapolation that estimates the pixels in a lost block by using the contractive mapping parameters of a range block homogeneous to the lost block. The proposed BLRA is applied to the lost blocks in the iteration of decoding.

With increasing Internet traffic congestion, the provision of reliable transmission and packet loss recovery continues to be of substantial importance. In this paper, we analyze a new recovery method using punctured convolutional codes, demonstrating the simplicity and efficiency of the proposed method for the recovery of lost packets. The analysis provides a method for determining the recoverability and the post-reconstruction receiving rate for a given convolutional code. The exact expressions for calculating the recovery rate are derived for a number of convolutional codes and the (2, 1, m) punctured convolutional code. Where packet loss probabilities are in the range typically found in Internet transmissions, the convolutional code-based method delivers superior performance over the traditional parity method with the same redundancy.

In this paper, we propose and analytically evaluate the use of punctured convolutional codes for recovering packets lost in multicast transmission. An independent erasure channel is assumed for packets transmission over a star topology. The analysis provides a method for determining the recoverability and the post-reconstruction receiving rate for a given convolutional code. We theoretically evaluate the effectiveness of the proposed approach taking into account two different parameters: the number of transmissions per packet and the number of packets needed to be sent to guarantee the reception of data. Finally, we compare the proposed approach with the scheme when parity packets are generated based on Reed-Solomon codes.