Packet loss

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

A new priority based congestion control protocol for Wireless Multimedia Sensor Networks

Abstract: New applications made possible by the rapid improvements and miniaturization in hardware has motivated recent developments in wireless multimedia sensor networks (WMSNs). As multimedia applications produce high volumes of data which require high transmission rates, multimedia traffic is usually high speed. This may cause congestion in the sensor nodes, leading to impairments in the quality of service (QoS) of multimedia applications. Thus, to meet the QoS requirements of multimedia applications, a reliable and fair transport protocol is mandatory. An important function of the transport layer in WMSNs is congestion control. In this paper, we present a new queue based congestion control protocol with priority support (QCCP-PS), using the queue length as an indication of congestion degree. The rate assignment to each traffic source is based on its priority index as well as its current congestion degree. Simulation results show that the proposed QCCP-PS protocol can detect congestion better than previous mechanisms. Furthermore it has a good achieved priority close to the ideal and near-zero packet loss probability, which make it an efficient congestion control protocol for multimedia traffic in WMSNs. As congestion wastes the scarce energy due to a large number of retransmissions and packet drops, the proposed QCCP-PS protocol can save energy at each node, given the reduced number of retransmissions and packet losses.

Apparatus and method for flow control

Abstract: In flow control apparatus formed of a network of switching hubs in a hierarchy configuration for performing data packet transfer within each of a plurality of respectively separate groups of terminals, based on use of group identifiers contained in the data packets, occurrence of congestion of the output section of a port of a switching hub is judged respectively separately for each of the terminal groups, and one or more congestion notification packets for effecting a pause in data packet transmission are generated and transmitted from that switching hub, directed only to one or more terminals of a group relating to the congestion. As a result, the effects of port congestion and of congestion control operations are limited to the terminal group concerned. Congestion can be judged for a terminal group based on a level of utilization of a port output buffer that is used only by that group, or based on a rate of flow of data from that group into a port output buffer which is used in common for all terminal groups, and congestion notification packets may be transmitted from a single port or from a plurality of ports of a switching hub, in accordance with the detected degree of congestion.

Low latency handling of transmission control protocol messages in a broadband satellite communications system

Abstract: An approach for transmitting packets conforming with the TCP (Transmission Control Protocol) over a satellite communications network comprises a plurality of prioritized queues that are configured to store the packets. The packets conform with a predetermined protocol. A classification logic classifies the packets based upon the predetermined protocol. The packet is selectively stored in one of the plurality of queues, wherein the one queue is of a relatively high priority. The packet is scheduled for transmission over the satellite communications network according to the relative priority of the one queue.

Flexible engine and data structure for packet header processing

Abstract: A pipelined linecard architecture for receiving, modifying, switching, buffering, queuing and dequeuing packets for transmission in a communications network. The linecard has two paths: the receive path, which carries packets into the switch device from the network, and the transmit path, which carries packets from the switch to the network. In the receive path, received packets are processed and switched in an asynchronous, multi-stage pipeline utilizing programmable data structures for fast table lookup and linked list traversal. The pipelined switch operates on several packets in parallel while determining each packet's routing destination. Once that determination is made, each packet is modified to contain new routing information as well as additional header data to help speed it through the switch. Each packet is then buffered and enqueued for transmission over the switching fabric to the linecard attached to the proper destination port. The destination linecard may be the same physical linecard as that receiving the inbound packet or a different physical linecard. The transmit path consists of a buffer/queuing circuit similar to that used in the receive path. Both enqueuing and dequeuing of packets is accomplished using CoS-based decision making apparatus and congestion avoidance and dequeue management hardware. The architecture of the present invention has the advantages of high throughput and the ability to rapidly implement new features and capabilities.

The effects of wide-area conditions on WWW server performance

Abstract: WWW workload generators are used to evaluate web server performance, and thus have a large impact on what performance optimizations are applied to servers. However, current benchmarks ignore a crucial component: how these servers perform in the environment in which they are intended to be used, namely the wide-area Internet.This paper shows how WAN conditions can affect WWW server performance. We examine these effects using an experimental test-bed which emulates WAN characteristics in a live setting, by introducing factors such as delay and packet loss in a controlled and reproducible fashion. We study how these factors interact with the host TCP implementation and what influence they have on web server performance. We demonstrate that when more realistic wide-area conditions are introduced, servers exhibit very different performance properties and scaling behaviors, which are not exposed by existing benchmarks running on LANs. We show that observed throughputs can give misleading information about server performance, and thus find that maximum throughput, or capacity, is a more useful metric. We find that packet losses can reduce server capacity by as much as 50 percent and increase response time as seen by the client. We show that using TCP SACK can reduce client response time, without reducing server capacity.