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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.


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Journal ArticleDOI
TL;DR: This research investigates the problem of observer-based robust H∞ output feedback control for networked control systems (NCSs) with randomly occurring uncertainties, dynamic quantization, and packet loss and designs a sufficient condition for the existence of such output feedback controller in the form of linear matrix inequality (LMI).
Abstract: This paper investigates the problem of observer-based robust H ∞ output feedback control for networked control systems (NCSs) with randomly occurring uncertainties, dynamic quantization, and packet loss. It is assumed that the system measurement output will be quantized by a dynamic quantizer and the random packet loss is represented by Bernoulli random binary distribution. In the presence of randomly occurring uncertainties, dynamic quantization, and packet loss, the attention of this research is focused on the design of observer-based robust H ∞ output feedback controller such that the resulting system is stochastically stable with the prescribed H ∞ noise attenuation level. The sufficient condition for the existence of such output feedback controller is expressed in the form of linear matrix inequality (LMI). Simulation results of two illustrative examples are proposed to demonstrate the effectiveness of the developed design method.

64 citations

Proceedings ArticleDOI
15 Nov 1999
TL;DR: This work presents a new protocol, UDP Lite, that provides the kind of protection often needed by these real time applications especially when they are run over wireless networks, and shows how UDP Lite uses a wireless network more efficiently for two different scenarios.
Abstract: Two rapidly evolving and merging technologies are the Internet and wireless data communications. Another trend is the increased use of real-time multimedia applications for audio and video communication. The most popular transport protocol among these delay sensitive applications is UDP (user datagram protocol), which is a lightweight protocol that provides multiplexing among user processes and has low protocol processing overhead. UDP's checksum policy is to protect either an entire packet including UDP and IP headers or nothing in the packet at all. This conforms badly with applications that prefers errors in the payload to the loss of whole packets. Many real-time applications fall into this category. We present a new protocol, UDP Lite, that provides the kind of protection often needed by these real time applications especially when they are run over wireless networks. UDP Lite increases the flexibility of UDP by providing an optionally partial checksum. Each packet can be optionally divided into a sensitive and an insensitive part by the sender. Errors in the sensitive part will cause packets to be discarded by the UDP Lite receiver, while errors in the insensitive part are ignored by UDP Lite. We show how UDP Lite uses a wireless network more efficiently for two different scenarios. By combining UDP Lite with compressed RTP, the gain can be even higher. Simulations show that the error rate of the network can increase by almost an order of magnitude without increasing the packet loss ratio. This enables cheaper network solutions for applications such as IP telephony.

63 citations

Proceedings ArticleDOI
23 Feb 2011
TL;DR: An in-depth experimental analysis of the use of HTTP-based request-response streams for video streaming, finding that request- response streams are able to scale with the available bandwidth by increasing the chunk size or the number of concurrent streams.
Abstract: Adaptive video streaming based on TCP/HTTP is becoming popular because of its ability to adapt to changing network conditions We present an in-depth experimental analysis of the use of HTTP-based request-response streams for video streaming In this scheme, video fragments are fetched by a client from the server, in smaller units called chunks, potentially via multiple parallel HTT P requests (TCP connections) A model for the achievable throughput is formulatedThe model is validated by a broad range of streaming experiments, including an evaluation of TCP-friendlinessOur findings include that request-response streams are able to scale with the available bandwidth by increasing the chunk size or the number of concurrent streams Several combinations of system parameters exhibiting TCP-friendliness are presented We also evaluate the video streaming performance in terms of video quality in the presence of packet loss Multiple request-response streams are able to maintain satisfactory performance, while a single TCP connection deteriorates rapidly with increasing packet loss The results provide experimental evidence that HTTP-based request-response streams are a good alternative to classical TCP streaming

63 citations

Proceedings ArticleDOI
Hyun-Ho Choi1, Osok Song1, Dong-Ho Cho2
29 Nov 2004
TL;DR: Numerical and simulation results show that the proposed handoff scheme performs well with respect to signaling cost, handoff delay, and packet loss compared with conventional schemes.
Abstract: In this paper, we present a practical UMTS-WLAN interworking architecture based on 3GPP standards, and propose a seamless handoff scheme that guarantees low delay and low packet loss during UMTS-WLAN handoff. For low handoff delay, the proposed handoff scheme performs pre-registration and pre-authentication processes before the layer 2 handoff. Moreover, it uses packet buffering and forwarding functions in order to reduce packet loss during the handoff period. Numerical and simulation results show that the proposed scheme performs well with respect to signaling cost, handoff delay, and packet loss compared with conventional schemes.

63 citations

Journal ArticleDOI
TL;DR: This paper proposes a mapping between some common QoS parameters such as latency and bit rate and the parameters used in the algorithm, and study the algorithm's performance and obtain simulation results for selected scenarios and configurations of interest.
Abstract: Bluetooth is a cable replacement technology for Wireless Personal Area Networks. It is designed to support a wide variety of applications such as voice, streamed audio and video, web browsing, printing, and file sharing, each imposing a number of quality of service constraints including packet loss, latency, delay variation, and throughput. In addition to QOS support, another challenge for Bluetooth stems from having to share the 2.4 GHz ISM band with other wireless devices such as IEEE 802.11. The main goal of this paper is to investigate the use of a dynamic scheduling algorithm that guarantees QoS while reducing the impact of interference. We propose a mapping between some common QoS parameters such as latency and bit rate and the parameters used in the algorithm. We study the algorithm's performance and obtain simulation results for selected scenarios and configurations of interest.

63 citations


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