Packet loss

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

User-Oriented QoS Analysis in MPEG-2 Video Delivery

Abstract: We address the problem of video quality prediction and control for high-resolution video transmitted over lossy packet networks. In packet video, the bitstream flows through several subsystems (coder, network, decoder); each of them can impair the information, either by data loss or by introducing some delay. However, each of these subsystems can be fine-tuned in order to minimize these problems and to optimize the quality of the delivered signal, taking into account the available bitrate. The assessment of the end-user quality is a non-trivial issue. We analyse how the user-perceived quality is related to the average encoding bitrate for variable bit rate MPEG-2 video. We then show why simple distortion metrics may lead to inconsistent interpretations. Furthermore, for a given coder setup, we analyse the effect of packet loss on the user-level quality. We then demonstrate that, when jointly studying the impact of coding bit rate and packet loss, the reachable quality is upperbound and exhibits one optimal coding rate for a given packet loss ratio.

Data path security processing

Abstract: Methods and associated systems provide secured data transmission over a data network. A security device provides security processing in the data path of a packet network. The device may include at least one network interface to send packets to and receive packets from a data network and at least one cryptographic engine for performing encryption, decryption and/or authentication operations. The device may be configured as an in-line security processor that processes packets that pass through the device as the packets are routed to/from the data network.

Method and system for transmission control protocol (TCP) packet loss recovery over a wireless link

Abstract: A method and system for Transmission Control Protocol (TCP) packet loss recovery over a wireless link where the packet loss is due to a high Bit Error Rate (BER). Such a method and system allows the exponential growth of a TCP congestion window for every acknowledgement packet that is received after a retransmission timers expires and/or three duplicate acknowledgement packets are received. In such a scenario, a maximum amount of data is transmitted via the exponentially growing TCP congestion window thereby optimizing and increasing packet throughput. A method and system for setting the congestion window to a greater amount than originally set of the maximum allowed window size is also presented. Further, a method and system for, as the BER changes between low and high, automatically adjusting the congestion window size to be a variable of the maximum allowed window size and dynamically adjusting the packet size is presented.

Study of information loss in packet voice systems

Abstract: Once a voice buffer is full, it remains full for a certain period, during which many packets are possibly blocked, resulting in consecutive clippings in voice. The packet loss rate during this period changes slowly and has large fluctuations. It is shown that the temporal behavior of packet loss, especially at high rate, is inherently determined by voice correlation and system capacity and is independent of buffer size. Buffering may reduce the occurrence of short blocking periods associated with low rates packet loss but does not affect long ones associated with high packet loss rates. In fact, increasing the buffer size merely extends nonblocking periods, and thereby reduces the overall average packet loss rate, but packet-loss performance within existing blocking periods is not significantly improved. A simple tool is developed for calculating the boundary performance. It is found that it is possible to design a packet-switched voice system without buffering only at the expense of supporting a fewer number of calls. The issue of voice delay allocation between source and network is discussed, and it is shown that it is more effective to keep the network delay short while extending the source delay. >

Global fairness of additive-increase and multiplicative-decrease with heterogeneous round-trip times

Abstract: Consider a network with an arbitrary topology and arbitrary communication delays, in which congestion control is based on additive-increase and multiplicative-decrease. We show that the source rates tend to be distributed in order to maximize an objective function called F/sub A//sup h/ ("F/sub A//sup h/ fairness"). We derive this result under the assumption of rate proportional negative feedback and for the regime of rare negative feedback. This applies to TCP in moderately loaded networks, and to those TCP implementations that are designed to interpret multiple packet losses within one RTT as a single congestion indication and do not rely on re-transmission timeout. This result provides some insight into the distribution of rates, and hence of packet loss ratios, which can be expected in a given network with a number of competing TCP or TCP-friendly sources. We validate our findings by analyzing a multiple-bottleneck scenario, and comparing with previous results (Floyd, 1991, Mathis et al, 1997) and an extensive numerical simulation with realistic parameter settings. We apply F/sub A//sup h/ fairness to gain a more accurate understanding of the bias of TCP against long round-trip times.