Retransmission Algorithms Research Articles

Joint Delay Reduction and Power of Device for Differentiated Data Transmission in D2D Network

In wireless communication, Device-to-Device (D2D) networks will become more and more common with the popularity of mobile terminals. However, in a D2D network, the processing capabilities of different terminals may be different, and the energy of different terminals is different due to battery life. In order to improve the quality of experience (QoE) of terminals with different processing capabilities and energy, and realize the transmission of differentiated data streams, this paper combines the existing base station retransmission and terminal retransmission algorithms in D2D network. In the transmission process, the data packet is divided into a basic layer and several enhancement layers according to the definition of the data. Based on the link quality, the basic layer data packet is transmitted firstly, then the enhancement layer data packet is transmitted. Moreover, not only the base station retransmits the data packet, but also the terminal that has received the corresponding data packet also participates in the retransmission. In order to extend the average life of the network, we propose the maximum weight packet selection algorithm based on channel quality and terminal energy (MWP-QE). The simulation results show that compared with the existing retransmission algorithm, this algorithm can realize the transmission of differentiated data, reduce the retransmission delay and extend the life of the network. A post-publication change was made to this article on 17 Jun 2020 to amend the author list.

A lightweight reliability mechanism proposed for datagram congestion control protocol over wireless multimedia sensor networks

AbstractTransmission of multimedia data over wireless multimedia sensor networks (WMSNs) is a challenging task and requires an effective design for the transport layer protocols to meet both the resource‐constrained WMSN and the requirements of multimedia streaming. The standard transport layer protocols (eg, UDP, TCP, SCTP, and DCCP) have been shown inefficient to transport multimedia data over WMSN without modification in the design of end‐to‐end (ETE) reliability and the congestion control mechanisms. This paper proposed a lightweight reliability mechanism for datagram congestion control protocol (LW‐DCCP) over WMSNs. The lightweight reliability mechanism consists of caching and retransmission algorithms that consider the resource‐constrained WMSNs and the requirements of video streaming over WMSN. Moreover, this paper presents an important modification to TCP‐friendly rate control congestion control mechanism that calculates the loss interval based on intracoded frame sending time. The LW‐DCCP has been evaluated through comprehensive simulation experiments and compared with baseline transport protocols using the 2 frameworks (Eval‐SCTP and Eval‐DCCP) that we developed in NS‐2 environment. The experimental results show that LW‐DCCP ensures effective congestion control and ETE reliability for video transmission over WMSN. In addition, LW‐DCCP experiences better performance in terms of energy consumption, ETE delay, jitter delay, frame delivery ratio, frame peak signal‐to‐noise ratio, and visual quality of the received video as compared to the baseline transport layer protocols.

An Improved TCP Congestion Control Algorithm of Based on Bandwidth Estimation in Heterogeneous Networks

—Due to the difficulty in discriminating the wireless loss from congestion loss in heterogeneous networks, TCP Westwood can not accurately estimate the available bandwidth, and cause low network utilization. In this paper, we propose an improved congestion control algorithm called TCP NewBR, which adaptively estimates bandwidth by using the bottleneck link utilization, and selects more precise time interval of ACK to improve the accuracy of available bandwidth estimation. In addition, we also modify the faster recovery and faster retransmission algorithms according to the queue length of bottleneck link. Simulation results in several scenarios show that TCP NewBR can get more accurate bandwidth estimation, and more significant throughput than traditional TCP Westwood, and fairer and friendlier than TCP Reno.

Improving fairness in IEEE 802.11 networks using MAC layer opportunistic retransmission

This paper introduces DAFMAC (Decode And Forward MAC), a scalable opportunistic cooperative retransmission enhancement for the IEEE 802.11 MAC protocol which operates without the need for additional explicit control signalling. Distributed opportunistic retransmission algorithms rely on selecting a single suitable relay without direct arbitration between nodes. Simulations show that DAFMAC offers a significant improvement in fairness for both throughput and jitter, giving multiple parallel data flows a more equal opportunity to utilise the channel. DAFMAC cooperative retransmissions are shown to reduce node energy consumption for a given throughput. Further, the DAFMAC relay selection algorithm is shown to scale very well in terms of complexity and memory requirements in comparison to other cooperative retransmission schemes.

Blind xor: Low-Overhead Loss Recovery for Vehicular Safety Communications

Packet loss in vehicular networks can undermine the effectiveness of communication-based accident prevention systems. Traditionally, retransmission has been used to combat packet loss in communication networks. However, retransmission overhead is known to increase channel congestion in dense wireless vehicular networks, which increases the likelihood of packet loss due to collisions, limiting the loss recovery capacity of the retransmission algorithm. This paper presents a low-overhead retransmission algorithm called blind xor (BXOR). By xoring multiple packets into a single retransmission, BXOR recovers an increased number of lost packets per retransmission. BXOR keeps the overhead low by not trying to learn the loss status of the receivers via feedback, rather, which packets to xor are blindly decided by the retransmitter alone. It is mathematically proved that BXOR can outperform existing retransmission methods if the conditional reception probability (CRP) of the xored packets is greater than 0.5. The performance is maximized by xoring an optimal number of packets, which is a function of the CRP. It is also proved that there is a negative performance if BXOR is exercised for a CRP of less than 0.5. Guided by the analytical results, we propose a practical BXOR protocol, which opportunistically exercises the xor operation based on the current estimation of the CRP. Simulation experiments confirm that, within a 110-m radius of the original packet transmitter, BXOR can reduce the packet reception failure rate by up to 60% of the rates achievable by previously proposed retransmission algorithms.

Effect of packet loss in video quality of experience

Today, packet video solutions are found everywhere, mainly due to the many advantages that they offer. Nevertheless, packet losses due to network issues like errors or jitter, affect decoder performance, thus generating video artifacts. The user's quality of experience decreases, depending among other factors, on the information carried by the lost packets. In this work we analyze the effect of packet loss in decoded video as a function of the part of the frame affected by the error and how the error is propagated because of temporal prediction. Then, we create a model which makes use of all the information available at the network level in a typical commercial multimedia deployment (such as Internet Protocol television [IPTV]). Finally, we apply that model to the overall design of content-aware network applications governing quality of service measures, video scrambling strategies, and smart retransmission algorithms.

DTN Bundle Layer over TCP: Retransmission Algorithms in the Presence of Channel Disruptions

“Challenged n etworks” often violate the TCP design key assumption of continuous path availability from the source to the sink node. These networks are the preferred target of Delay/Disruption Tolerant Networking (DTN), which aims at providing a more robust network architecture. Against disruptions, however, even standard TCP offers a certain level of robustness, through its many retransmission algorithms. This intrinsic TCP resilience against disruptions is retained and enhanced if TCP is inserted in a DTN architecture. Focusing on this, the aim of the paper is to present an in-depth analysis of TCP and DTN bundle protocol retransmission algorithms that are triggered by channel disruptions, thus providing the reader with a comprehensive view of the many mechanisms involved and their complex interactions. The first sections, devoted to the description of TCP and DTN algorithms, are completed by the presentation of some numerical evaluations obtained by means of a Linux testbed. They refer to a GEO satellite environment and clarify the previous algorithm descriptions. Moreover, they also offer the reader some useful insights on both TCP and DTN resilience to disruptions, considering some performance metrics, like the “ maximum tolerable disrupti on length” and the “restart delay”, introduced and explained in the paper.

A MAC-Layer Retransmission Algorithm Designed for the Physical-Layer Characteristics of Clustered Sensor Networks

We consider scenarios in which an event, such as a mobile node arriving to collect data or a data-request arriving at a node via the sensor network, initiates the collection of one packet of data from each node in the surrounding cluster of nodes. The node requesting the data broadcasts its identity and purpose to the cluster and assumes the role of clusterhead. This broadcast message is also used to synchronize all nodes in the cluster, which covers an area with a radius of several hundred meters, to within a few microseconds. We exploit this ability to synchronize the cluster to improve retransmission algorithms in CSMA/CA-based MAC protocols. The result is a retransmission algorithm that uses a synchronized, shared contention window (SSCW) that is easy to implement and results in fewer collisions than retransmission algorithms that use binary-exponential-backoff (BEB). A simple analytical approach is further proposed to optimize the size of the shared contention window. We show via analytical, numerical, and simulation results that an SSCW-based 802.11 DCF protocol performs significantly better in these scenarios we consider than both 802.15.4 (ZigBee) and BEB-based 802.11 DCF (WiFi). We also show that SSCW-based 802.11 is more fault-tolerant and fair than TDMA