A dynamic partial buffer sharing scheme for packet loss control in congested networks

To provide the differential quality of service (QoS) for different classes of packets and reduce the packet loss probability (PLP), a novel priority-based composite assembly scheme for optical burst switching (OBS) networks is proposed. The low and high packet classes are aggregated into a single burst simultaneously, and the highest-priority packets are placed in the middle, while the low-priority packets are at the tail and head of the burst. The priority is lowered gradually from the middle to the ends. Simulation results demonstrate that the proposed assembly strategy not only guarantees the integrity of the high-priority bursts, but also significantly reduces the average end-to-end delay of the bursts and the PLP of network. So it can adapt to the flexible network with QoS requirement.

A typical operating environment of a packet switching (store-and-forward) computer communication network is that it is shared by many users with different classes of packets. Packets may be classified in a very general fashion by types of users, messages, applications, transactions, response time requirements, packet parameters such as packet rate and length, and by network parameters such as source-destination and path length. A well-designed network must provide access and performance assurance to all packet classes. This paper presents a heuristic algorithm for designing such a communication network. The algorithm presented contains heuristic algorithms for discrete link capacity assignment, priority assignment, and flow assignment problems with an additional feature which allows one to alter network topology interactively. Sample results from applications of the overall network design are also given.

In this paper, the consensus problem is studied for a class of multi-agent systems with sampled data and packet losses, where random and deterministic packet losses are considered, respectively. For random packet losses, a Bernoulli-distributed white sequence is used to describe packet dropouts among agents in a stochastic way. For deterministic packet losses, a switched system with stable and unstable subsystems is employed to model packet dropouts in a deterministic way. The purpose of this paper is to derive consensus criteria, such that linear multi-agent systems with sampled-data and packet losses can reach consensus. By means of the Lyapunov function approach and the decomposition method, the design problem of a distributed controller is solved in terms of convex optimization. The interplay among the allowable bound of the sampling interval, the probability of random packet losses, and the rate of deterministic packet losses are explicitly derived to characterize consensus conditions. The obtained criteria are closely related to the maximum eigenvalue of the Laplacian matrix versus the second minimum eigenvalue of the Laplacian matrix, which reveals the intrinsic effect of communication topologies on consensus performance. Finally, simulations are given to show the effectiveness of the proposed results.In this paper, the consensus problem is studied for a class of multi-agent systems with sampled data and packet losses, where random and deterministic packet losses are considered, respectively. For random packet losses, a Bernoulli-distributed white sequence is used to describe packet dropouts among agents in a stochastic way. For deterministic packet losses, a switched system with stable and unstable subsystems is employed to model packet dropouts in a deterministic way. The purpose of this paper is to derive consensus criteria, such that linear multi-agent systems with sampled-data and packet losses can reach consensus. By means of the Lyapunov function approach and the decomposition method, the design problem of a distributed controller is solved in terms of convex optimization. The interplay among the allowable bound of the sampling interval, the probability of random packet losses, and the rate of deterministic packet losses are explicitly derived to characterize consensus conditions. The obtained criteria are closely related to the maximum eigenvalue of the Laplacian matrix versus the second minimum eigenvalue of the Laplacian matrix, which reveals the intrinsic effect of communication topologies on consensus performance. Finally, simulations are given to show the effectiveness of the proposed results.

The transmission of video traffic over the Internet is a fundamental issue of network design. Video applications require quality of service guarantees from the network in terms of limited packet loss, end-to-end delay, and delay variation. The question of today’s research and development is how to provide these guarantees considering the architecture of the present Internet. In the last years a variety of admission control schemes based on per-hop or end-to-end measurements has been suggested to control delay and loss sensitive streams with very little or no support at the routers. Most of these solutions, however, have to apply the same acceptance threshold for all streams, a significant limitation considering the diverse quality requirements of the applications. In this work we investigate how source shaping and forward error correction (FEC) can be used together to achieve application specific quality differentiation in terms of end-to-end delay and packet loss probability. While source shaping and FEC have been proposed independently to decrease the probability of packet loss due to buffer overflow, their joint use has not been studied before.As the two control functions use the same scarce resource, end-node delay, and their efficiency to decrease loss probability is proportional to the introduced delay but with a decreasing marginal gain, combining the two a better performance can be achieved than by using only one of them.The performance evaluation focuses on the optimal delay allocation for shaping and FEC, such that the loss probability is minimized. We investigate how shaping can be used to substitute FEC redundancy and the sensitivity of the quality differentiation to the background traffic characteristics.KeywordsQuality of Servicesource shapingFEC

Accurate modeling of VoIP traffic QoS parameters in current and future networks with multifractal and Markov models

The subject of research in the article is the processes of fast rerouting with the protection of the Quality of Service level in infocommunication networks. The aim of the work is the development of a mathematical model of Fast ReRouting with the protection of the Quality of Service level by the bandwidth and probability of packet loss for data traffic. The following tasks are solved in the article: development and research of a fast rerouting flow-based model with the protection of the Quality of Service level of data traffic in the infocommunication network. The following methods are used: the graph theory, the teletraffic theory, the queuing theory, and mathematical programming methods. The following results were obtained: the fast rerouting flow-based model was developed and investigated, which, due to the introduced protection conditions on indicators of bandwidth and packet loss probability, allows providing the Quality of Service along both the primary and backup multipath. Conclusions: Within the framework of the proposed flow-based model, the fast rerouting technological task was formulated as an optimization problem with the constraints of the conditions of implementation of the multipath routing strategy, conditions of flow conservation, and conditions of protection of the link, node, and level of Quality of Service in terms of bandwidth and packet loss probability. The application of this solution contributes to the optimal use of the available network resource while providing the specified level of Quality of Service in terms of bandwidth and probability of packet loss along both the primary and backup routes in the case of a failure of the infocommunication network elements. The proposed flow-based model can be used as the basis of the algorithmic software of existing routers and/or controllers of Software-Defined Networks, which are responsible for the calculation of the primary and backup paths in the fast rerouting of data traffic sensitive to such Quality of Service indicators as bandwidth and packet loss probability.

Time synchronization is critical for time-sensitive cell-free massive MIMO systems, as packet loss from mobility or poor channels degrades performance. In this paper, we propose robust coordinated time synchronization (Co-TS) schemes for the cell-free massive MIMO to combat the impact of packet loss. Specifically, a multi-criteria based AP association mechanism is first proposed where in addition to the large-scale fading factor, the effects of received signal strength and packet loss probability are also considered during the AP-device association process. In addition, the fuzzy logic theory is utilized to balance among the three criteria to achieve higher robustness against packet loss over the conventional single-criterion based AP association approaches. After that, a Co-TS model considering both the coordination among APs and the effect of packet loss is developed, where the unsynchronized node can receive a set of timestamps from its associated APs during each synchronization cycle. We use maximum likelihood estimation (MLE) to jointly estimate the clock offset and skew under Gaussian-distributed random delays, and derive their closed-form Cramér-Rao Lower Bounds. Simulation results demonstrate that our proposed schemes are robust against packet loss and can significantly enhance time synchronization performance over the existing methods.

In recent years, wavelet transforms have been successfully applied to image and video compression. In a distributed, multimedia computing environment, compressed video and image files need to be transmitted to users for real-time replay. Because the network is subject to packet losses, the image packetization scheme should take packet losses into account in their design. In this paper, we present an error concealment algorithm for wavelet-based compressed images over packet-switched networks. Two different packetization schemes, called the intrablock-oriented and interblock-oriented schemes, in conjunction with wavelet coding, are presented. In our approach, adjacent wavelet coefficients are scattered into different packets, so that any packet contains encoded wavelet coefficients at different resolution layers. When a packet is lost due to network congestion or transmission errors, a lost wavelet coefficient is estimated by the receiver through a linear interpolation of its adjacent ones. Our approach is evaluated under two different packet loss models with various packet loss probabilities through simulations driven by actual video sequences.© (1996) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

The high mobility of trains and complex line environment will lead to poor communication quality, which will seriously affect the stability of the virtually coupled train set (VCTS). Packet loss is one of the important indexes to evaluate communication quality. In this paper, a numerical method is proposed to analyze the influence of random packet loss on the stability of the homogeneous VCTS. Firstly, combining with the characteristics of packet loss design probability model. Based on the longitudinal dynamics model and the constraints of the train interval distance, a train operation control model is formulated with considering random packet loss. Secondly, the stability conditions of the homogeneous VCTS expressed by packet loss probability are derived from the perspective of frequency domain. Finally, the effects of random packet loss on the stability of the homogeneous VCTS are analyzed and the validity of the theoretical results is confirmed via numerical simulations.

LQG-like control of scalar systems over communication channels: The role of data losses, delays and SNR limitations

When the number of packets increases beyond the network capacity leads to congestion in the computer network. Congestion effects the QOS parameters severely by decreasing throughput and increasing packet loss and delay, thus degrading the overall network performance. Active queue management algorithm is router based mechanism for early detection of congestion in the computer network. The basic idea of AQM is to sense and detect congestion in advance and to inform the sender to reduce its sending rate there by reducing the number of packets sent in the network to control the congestion in the CN. AQM algorithm is congestion control algorithm which increases throughput, link utilization and decreases the delay and packet loss. RED (Random Early Detection) provides the mechanism for congestion avoidance. The technique has several drawbacks and one among them is inability to deal with busty traffic. The major drawback of Blue algorithm is that it detect the congestion by analyzing and observing the queue length, queue length may little information about the severity of the congestion and detection of number of competing connections. The proposed algorithm called AdvancedBlue which is the modification of Blue AQM algorithm uses idle link, packet loss to detect congestion decouples the congestion measure with performance measure and stabilizes the performance measure against the target independently of number of users. AdvancedBlue AQM is simulated in wireless network using NS2 simulator. Wireless network is quite different from the wired network. The nodes have no wired links to other nodes and base station. So any barrier that is between the sender and the receiver could influence the transmission power and bring packet loss. The proposed AQM showed a better performance in wired network and was more complicated in wireless network. The proposed algorithm when compared with existing DropTail algorithm in wireless network showed better performance with respect to throughput and delay but had more packet loss. Keywords: AQM, AdvancedBlue, Blue, CN, DropTail, NS2, QOS, RED Cite this Article: Uma R. Pujeri. Advanced AQM Blue Algorithm in Wireless Network. International Journal of Wireless Network Security. 2020; 6(1): 16–24p.

Packet loss probability alone does not provide a complete characterization of the quality of a network's real-time packet sessions. Additional performance measures such as the consecutive packet loss (the so called average packet gap) must be considered. The authors propose to improve the packet gap performance of asynchronous transfer mode (ATM) sessions by minimizing the packet gap present in the network's buffers. They introduce a simple packet dropping policy that, in steady state, minimizes: (i) the packet gap, and (ii) the packet buffering delay, when this delay can be reduced without increasing the packet gap, or packet loss probability.

A backpropagation neural network cIassifier is used for distinguishing network congestion and link error over ad hoc networks. TCP considers a 11 packet losses as network congestion, and reacts to such congestion by reducing its data rate. In ad hoc networks, there are many packet losses are due to Iink errors, but not due to network congestion; and then TCP often reduces its data rate mistakenly, cannot keep a reasonable data rate. Then we introduce an improvement TCP with a packet loss classifier, TCP-BP algorithm. The result of our simulation shows that the TCP-BP algorithm is superior to Vegas and TCP Westwood algorithm in TCP throughput.

Normally sensor nodes are located at unattended environment which has low battery power. The wireless hypermedia sensor based networks have huge capacity and continuous flow of data from source to sink. Congestion arises at the time of incoming data is greater than available resources. The congestion causes packet loss, buffer over flow, energy wastage and increases end to end delay. The proposed method Dynamic Alternate Buffer switching and Congestion Control is used to manage congestion in an efficient manner. In which congestion is detected based on remaining buffer, remaining energy and trust level of the sensor nodes. This approach is based on cost evaluation method which decides primary and spare buffers at run time. This method is used to enable dynamic buffer switching and optimize the effect of congestion. The performance of BETCC compared with various existing protocol like TCEER and TFCC in terms of energy consumption and data loss ratio.

This paper proposes a novel error-resilient packet video coding method which is designed to operate on unreliable IP networks suffering from large bursts of packet losses. The main novelties of the proposed system are: (1) A Bayesian packet loss predictor is employed to dynamically predict the packet loss probabilities; (2) A harmonic frame-expansion is applied to the video stream to add error-resilience. This includes dynamically choosing the size of the expansions. (3) Error-propagation is reduced by filtering out error-prone high-frequency components from reference images. This means that the proposed video codec method does not need to switch between the inter and intra coding modes. (4) Theoretical analysis of the video performance is given. Our preliminary simulations have shown that the frame expansion method can provide excellent resilience to video packet losses. The system has generated high quality videos at low bit-rates (64kbs) on simulated networks with the packet loss probability ranging from 5% to 20%. Visual inspection has shown that under similar PSNRs, the resulting videos have noticeable less visual artifacts as compared with those from the conventional codecs