Packet loss

About: Packet loss is a research topic. Over the lifetime, 21235 publications have been published within this topic receiving 302453 citations.

Resource Allocation and Quality of Service Evaluation for Wireless Communication Systems Using Fluid Models

Abstract: Wireless systems offer a unique mixture of connectivity, flexibility, and freedom. It is therefore not surprising that wireless technology is being embraced with increasing vigor. For real-time applications, user satisfaction is closely linked to quantities such as queue length, packet loss probability, and delay. System performance is therefore related to, not only Shannon capacity, but also quality of service (QoS) requirements. This work studies the problem of resource allocation in the context of stringent QoS constraints. The joint impact of spectral bandwidth, power, and code rate is considered. Analytical expressions for the probability of buffer overflow, its associated exponential decay rate, and the effective capacity are obtained. Fundamental performance limits for Markov wireless channel models are identified. It is found that, even with an unlimited power and spectral bandwidth budget, only a finite arrival rate can be supported for a QoS constraint defined in terms of exponential decay rate

Admission policies for integrated voice and data traffic in CDMA packet radio networks

Abstract: In this paper, we derive optimal admission policies for integrated voice and data traffic in packet radio networks employing code division multiple access (CDMA) with directsequence spread spectrum (DS/SS) signaling. The network performance is measured in terms of the average blocking probability of voice calls and the average delay and packet loss probability of data messages. Our admission scheme determines the number of newly arrived voice users that are accepted in the network so that the long-term blocking probability of voice calls is minimized. In addition, new data arrivals are rejected if the mean delay or the packet loss probability of data exceeds a desirable prespecified level. A semi-Markov decision process (SMDP) is used to model the system operation

Low-complexity, low-delay, scalable and embedded speech and audio coding with adaptive frame loss concealment

Abstract: High-quality, low-complexity and low-delay scalable and embedded system and method are disclosed for coding speech and general audio signals. The invention is particularly suitable in Internet Protocol (IP)-based multimedia communications. Adaptive transform coding, such as a Modified Discrete Cosine Transform, is used, with multiple small-size transforms in a given signal frame to reduce the coding delay and computational complexity. In a preferred embodiment, for a chosen sampling rate of the input signal, one or more output sampling rates may be decoded with varying degrees of complexity. Multiple sampling rates and bit rates are supported due to the scalable and embedded coding approach underlying the present invention. Further, a novel adaptive frame loss concealment approach is used to reduce the distortion caused by packet loss in communications using IP networks.

GI/Geom/1 queue based on communication model for mesh networks

Abstract: In mesh networks architecture, it should be permitted to visit the mobile client points. Whereas in mesh networks environment, the main throughput flows usually communicate with the conventional wired network. The so-called gateway nodes can link directly to traditional Ethernet, depending on these mesh nodes, and can obtain access to data sources that are related to the Ethernet. In wireless mesh networks WMNs, the quantities of gateways are limited. The packet-processing ability of settled wireless nodes is limited. Consequently, throughput loads of mesh nodes highly affect the network performance. In this paper, we propose a queuing system that relied on traffic model for WMNs. On the basis of the intelligent adaptivenes, the model considers the influences of interference. Using this intelligent model, service stations with boundless capacity are defined as between gateway and common nodes based on the largest hop count from the gateways, whereas the other nodes are modeled as service stations with certain capacity. Afterwards, we analyze the network throughput, mean packet loss ratio, and packet delay on each hop node with the adaptive model proposed. Simulations show that the intelligent and adaptive model presented is precise in modeling the features of traffic loads in WMNs. Copyright © 2013 John Wiley & Sons, Ltd.

Rate control for streaming video over wireless

Abstract: Rate control is an important issue in video streaming applications for both wired and wireless networks. A widely accepted rate control method in wired networks is equation based rate control (Sally Floyd et al., Aug. 2000), in which the TCP friendly rate is determined as a function of packet loss rate, round trip time and packet size. This approach, also known as TFRC, assumes that packet loss in wired networks is primarily due to congestion, and as such is not applicable to wireless networks in which the bulk of packet loss is due to error at the physical layer. We propose multiple TFRC connections as an end-to-end rate control solution for wireless video streaming. We show that this approach not only avoids modifications to the network infrastructure or network protocol, hut also results in full utilization of the wireless channel. NS-2 simulations and experiments over 1/spl times/RTT CDMA wireless data network are carried out to validate, and characterize the performance of our proposed approach.