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Packet loss

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


Papers
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Patent
George P. Su1, John K. Murata1
03 Nov 2006
Abstract: A method and apparatus for the dynamic adjustments of video streams sent from a server to a client is provided. Using both short term and long term thinning mechanisms, the server reduces its data transfer rate when bandwidth is reduced and increases its data transfer rate when bandwidth is available. In the short term mechanism, the server will dynamically adjust the transmission rate of the video or the quality of the video based on parameters returned by the client. These parameters include currently-buffered data-size, buffered-data duration, and packet loss rate. In the long term mechanism, the server will adjust the current maximum quality level allowed depending upon the round trip times of data packets transmitted. This invention decreases client rebuffering and packet loss and increases the efficiency of available bandwidth.

67 citations

Journal ArticleDOI
TL;DR: An optical flat datacenter network based on scalable optical switch system with optical flow control with RF tone in-band labeling technique allowing parallel processing of the label bits ensures the low latency operation regardless of the switch port-count.
Abstract: We propose and demonstrate an optical flat datacenter network based on scalable optical switch system with optical flow control. Modular structure with distributed control results in port-count independent optical switch reconfiguration time. RF tone in-band labeling technique allowing parallel processing of the label bits ensures the low latency operation regardless of the switch port-count. Hardware flow control is conducted at optical level by re-using the label wavelength without occupying extra bandwidth, space, and network resources which further improves the performance of latency within a simple structure. Dynamic switching including multicasting operation is validated for a 4x4 system. Error free operation of 40 Gb/s data packets has been achieved with only 1 dB penalty. The system could handle an input load up to 0.5 providing a packet loss lower that 10−5 and an average latency less that 500ns when a buffer size of 16 packets is employed. Investigation on scalability also indicates that the proposed system could potentially scale up to large port count with limited power penalty.

67 citations

Journal ArticleDOI
TL;DR: In this article, a dynamic rate control mechanism for voice and video traffic is proposed to achieve better statistical gain for voice traffic and to relieve congestion in fast packet networks, where the feedback delay for the source node to obtain the network congestion information is represented in the model.
Abstract: To achieve better statistical gain for voice and video traffic and to relieve congestion in fast packet networks, a dynamic rate control mechanism is proposed. An analytical model is developed to evaluate the performance of this control mechanism for voice traffic. The feedback delay for the source node to obtain the network congestion information is represented in the model. The study indicates that significant improvement in statistical gain can be realized for smaller capacity links (e.g., links that can accommodate less than 24 voice calls) with a reasonable feedback time (about 100 ms). The tradeoff for increasing the statistical gain is temporary degradation of voice quality to a lower rate. It is shown that whether the feedback delay is exponentially distributed or constant does not significantly affect performance in terms of fractional packet loss and average received coding rate. It is also shown that using the number of calls in talkspurt or the packet queue length as measures of congestion provides comparable performance. >

67 citations

Patent
24 Mar 1994
TL;DR: In this paper, the adaptive packet stream transmission mechanism is proposed, which includes modulo-2 sum circuitry for deriving a parity packet for each L transmitted data packets and transmission control circuitry selectively enables transmission of parity packet after each L data packets have been transmitted.
Abstract: A packet communication system transmits a stream of L data packets with at least one parity packet transmitted as an L+1th packet. The parity packet comprises at least, N data segments, each ith data segment being the modulo-2 sum of all identically positioned data segments in the stream of L packets. A node includes adaptive packet stream transmission apparatus which comprises a packet queue for holding a series of packets ready for transmission to a destination node. The adaptive packet stream transmission mechanism further includes modulo-2 sum circuitry for deriving a parity packet for each L transmitted data packets. Transmission control circuitry selectively enables transmission of a parity packet after each L data packets have been transmitted. A processor in the node controls the transmission control circuitry to vary the value of L in accordance with a determined network metric. The network metric may indicate a level of network congestion, with the processor increasing the value of L if the network metric manifests a lessened network congestion, and vice versa. The processor is also adapted to vary the value of L in accordance with a data error rate metric.

67 citations

Proceedings ArticleDOI
05 Jun 2000
TL;DR: This work looks at the issues involved in performing robust speech recognition over a packet-based network such as the IP network and a novel detection and estimation scheme for missing frames is introduced to overcome this problem.
Abstract: This work looks at the issues involved in performing robust speech recognition over a packet-based network such as the IP network. This involves the combination of robust speech recognition together with a reliable method of sending speech data over the IP network. The format in which the speech is sent over the network is considered and results show that much better robustness is achieved when the front-end features are transmitted directly rather than encoding the speech with a codec. The problem of packet loss is addressed and a novel detection and estimation scheme for missing frames is introduced to overcome this problem. This is shown to recover performance with 50% packet loss from 33% to 90% which is only 3% below the no loss case.

67 citations


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Performance
Metrics
No. of papers in the topic in previous years
YearPapers
2023133
2022325
2021694
2020846
20191,033
2018993