• Title/Summary/Keyword: adaptive bandwidth.

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Channel Interference Analysis of Wideband WLAN Based IEEE802.11n for 3rd Generation Digital Signage (3세대 디지털 사이니지를 위한 IEEE802.11n 광대역 무선랜에 대한 채널 간섭 분석)

  • Ko, Hojeong
    • Journal of Satellite, Information and Communications
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    • v.11 no.1
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    • pp.6-11
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    • 2016
  • In this paper, we have analyzed the effects of co-channel, adjacent-channel, and the human shield(Body Blockage) for wideband WLAN based on the IEEE802.11n 40MHz channel bandwidth required for high speed digital signage service. Simulation results show that wideband WLAN can be operated with 78 interferers over 63m distance in co- channel, 80 interferer over 61m distance in adjacent channel. By applying the mitigation method for reducing the interference, we have confirmed that protection distance is improved to 51m using beamforming, and 40m using cognitive radio in co-channel interference. Also body blockage interference is reduced using adaptive channel bandwidth, C/I ratio, beamforming, power control mitigation methodology.

An Adaptive FEC based Error Control Algorithm for VoIP (VoIP를 위한 적응적 FEC 기반 에러 제어 알고리즘)

  • Choe, Tae-Uk;Jeong, Gi-Dong
    • The KIPS Transactions:PartC
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    • v.9C no.3
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    • pp.375-384
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    • 2002
  • In the current Internet, the QoS of interactive applications is hardly guaranteed because of variable bandwidth, packet loss and delay. Moreover, VoIP which is becoming an important part of the information infra-structure in these days, is susceptible to network packet loss and end-to-end delay. Therefore, it needs error control mechanisms in network level or application level. The FEC-based error control mechanisms are used for interactive audio application such as VoIP. The FEC sends a main information along with redundant information to recover the lost packets and adjusts redundant information depending on network conditions to reduce the bandwidth overhead. However, because most of the error control mechanisms do not consider end-to-end delay but packet loss rate, their performances are poor. In this paper, we propose a new error control algorithm, SCCRP, considering packet loss rate as well as end-to-end delay. Through experiments, we confirm that the SCCRP has a lower packet loss rate and a lower end-to-end delay after reconstruction.

A Phase Locked Loop with Resistance and Capacitance Scaling Scheme (저항 및 커패시턴스 스케일링 구조를 이용한 위상고정루프)

  • Song, Youn-Gui;Choi, Young-Shig;Ryu, Ji-Goo
    • Journal of the Institute of Electronics Engineers of Korea SD
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    • v.46 no.4
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    • pp.37-44
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    • 2009
  • A novel phase-locked loop(PLL) architecture with resistance and capacitance scaling scheme has been proposed. The proposed PLL has three charge pumps. The effective capacitance and resistance of the loop filter can be scaled up/down according to the locking status by controlling the direction and magnitude of each charge pump current. This architecture makes it possible to have a narrow bandwidth and low resistance in the loop filter, which improves phase noise and reference spur characteristics. It has been fabricated with a 3.3V $0.35{\mu}m$ CMOS process. The measured locking time is $25{\mu}s$ with the measured phase noise of -105.37 dBc/Hz @1MHz and the reference spur of -50dBc at 851.2MHz output frequency

Hybrid Link State Update Algorithm in QoS Routing (하이브리드 QoS 라우팅 링크 상태 갱신 기법)

  • Cho, Kang Hong
    • Journal of the Korea Society of Computer and Information
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    • v.19 no.3
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    • pp.55-62
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    • 2014
  • This paper has proposed Hybrid QoS Routing Link State Update(LSU) Algorithm that has had a both advantage of LSU message control in periodic QoS routing LSU algorithm and QoS routing performance in adaptive LSU algorithm. Hybrid LSU algorithm can adapt the threshold based network traffic information and has the mechanism that calculate LSU message transmission priority using the flow of statistical request bandwidth and available bandwidth and determine the transmission of the message according to update rate per a unit of time. We have evaluated the performance of the proposed algorithm and the existing algorithms on MCI simulation network using the performance metric as the QoS routing blocking rate and the mean update rate per link, it thus appears that we have verified the performance of this algorithm that it can diminish to 10% of the LSU message count.

An Adaptive Buffer Tuning Mechanism for striped transport layer connection on multi-homed mobile host (멀티홈 모바일 호스트상에서 스트라이핑 전송계층 연결을 위한 적응형 버퍼튜닝기법)

  • Khan, Faraz-Idris;Huh, Eui-Nam
    • Journal of Internet Computing and Services
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    • v.10 no.4
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    • pp.199-211
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    • 2009
  • Recent advancements in wireless networks have enabled support for mobile applications to transfer data over heterogeneous wireless paths in parallel using data striping technique [2]. Traditionally, high performance data transfer requires tuning of multiple TCP sockets, at sender's end, based on bandwidth delay product (BDP). Moreover, traditional techniques like Automatic TCP Buffer Tuning (ATBT), which balance memory and fulfill network demand, is designed for wired infrastructure assuming single flow on a single socket. Hence, in this paper we propose a buffer tuning technique at senders end designed to ensure high performance data transfer by striping data at transport layer across heterogeneous wireless paths. Our mechanism has the capability to become a resource management system for transport layer connections running on multi-homed mobile host supporting features for wireless link i.e. mobility, bandwidth fluctuations, link level losses. We show that our proposed mechanism performs better than ATBT, in efficiently utilizing memory and achieving aggregate throughput.

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Computational Analytics of Client Awareness for Mobile Application Offloading with Cloud Migration

  • Nandhini, Uma;TamilSelvan, Latha
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.8 no.11
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    • pp.3916-3936
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    • 2014
  • Smartphone applications like games, image processing, e-commerce and social networking are gaining exponential growth, with the ubiquity of cellular services. This demands increased computational power and storage from mobile devices with a sufficiently high bandwidth for mobile internet service. But mobile nodes are highly constrained in the processing and storage, along with the battery power, which further restrains their dependability. Adopting the unlimited storage and computing power offered by cloud servers, it is possible to overcome and turn these issues into a favorable opportunity for the growth of mobile cloud computing. As the mobile internet data traffic is predicted to grow at the rate of around 65 percent yearly, even advanced services like 3G and 4G for mobile communication will fail to accommodate such exponential growth of data. On the other hand, developers extend popular applications with high end graphics leading to smart phones, manufactured with multicore processors and graphics processing units making them unaffordable. Therefore, to address the need of resource constrained mobile nodes and bandwidth constrained cellular networks, the computations can be migrated to resourceful servers connected to cloud. The server now acts as a bridge that should enable the participating mobile nodes to offload their computations through Wi-Fi directly to the virtualized server. Our proposed model enables an on-demand service offloading with a decision support system that identifies the capabilities of the client's hardware and software resources in judging the requirements for offloading. Further, the node's location, context and security capabilities are estimated to facilitate adaptive migration.

The Design of Terrestrial DMB Media Processor for Multi-Channel Audio Services (멀티채널 오디오 서비스를 위한 지상파 DMB 미디어처리기 설계)

  • Kang Kyeongok;Hong Jaegeun;Seo Jeongil
    • The Journal of the Acoustical Society of Korea
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    • v.24 no.4
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    • pp.186-193
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    • 2005
  • The Terrestrial Digital Multimedia Broadcasting (T-DMB) system supplies high quality audio comparable with VCD in 7 inch display and high quality audio comparable CD at the mobile reception environment T-DMB will launch commercial service at the middle of 2005. However the bandwidth for audio data and the number of channels are restricted to 128 kbps and 2 respectively in the current T-DMB standard because of the limitation of available bandwidth for multimedia data. This Paper Proposes a novel media processor structure for providing multi-channel audio contents oyer T-DMB system allowing backward compatibility with the legacy T-DMB receiver. Furthermore. we also Propose an adaptive receiver structure to supply optimal audio contents on various speaker configuration in T-DMB receiver. To provide multi-channel audio contents allowing backward comaptilbity with the legacy T-DMB receiver, the additional data for multi-channel audio are defined as a dependent stream of main audio stream. The OD strucure for control an additional multi-channel audio elementary stream is proposed without changing the BIFS of the legacy T-DMB system.

A Symbol Timing Recovery scheme using the jitter mean of adaptive loop filter in ATSC DTV systems (적응적 루프필터의 지터 평균값을 이용한 ATSC DTV 심볼 타이밍 동기 방식)

  • Kim, Joo-Kyoung;Lee, Joo-hyoung;Song, Hyun-keun;Kim, Jae-Moung;Kim, Seung-Won
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.42 no.10 s.340
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    • pp.1-8
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    • 2005
  • This Paper Proposes the algorithm for improving the Performance or symbol timing synchronization in hoc terrestrial DTV system. The Gardner algerian is used for symbol timing synchronization has good performance in multipath fading environment but degradation of performance is caused by jitter. Though the amount of jitter becomes more little as narrow bandwidth of loop Inter, convergence speed becomes slower. This paper propose the algorithm that averages out values of loop filter every certain time and gradually reduces the bandwidth of loop filter after estimating offset using this average for the high speed of convergence and reducing the met of jitter. The proposed algorithm has better performance with high speed of convergence and the amount of jitter than conventional method.

Adaptive Mitigation of Narrowband Interference in Impulse Radio UWB Systems Using Time-Hopping Sequence Design

  • Khedr, Mohamed E.;El-Helw, Amr;Afifi, Mohamed Hossam
    • Journal of Communications and Networks
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    • v.17 no.6
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    • pp.622-633
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    • 2015
  • The coexistence among different systems is a major problem in communications. Mutual interference between different systems should be analyzed and mitigated before their deployment. The paper focuses on two aspects that have an impact on the system performance. First, the coexistence analysis, i.e. evaluating the mutual interference. Second aspect is the coexistence techniques, i.e. appropriate system modifications that guarantee the simultaneous use of the spectrum by different technologies. In particular, the coexistence problem is analyzed between ultra-wide bandwidth (UWB) and narrow bandwidth (NB) systems emphasizing the role of spectrum sensing to identify and classify the NB interferers that mostly affect the performance of UWB system. A direct sequence (DS)-time hopping (TH) code design technique is used to mitigate the identified NB interference. Due to the severe effect of Narrowband Interference on UWB communications, we propose an UWB transceiver that utilizes spectrum-sensing techniques together with mitigation techniques. The proposed transceiver improves both the UWB and NB systems performance by adaptively reducing the mutual interference. Detection and avoidance method is used where spectrum is sensed every time duration to detect the NB interferer's frequency location and power avoiding it's effect by using the appropriate mitigation technique. Two scenarios are presented to identify, classify, and mitigate NB interferers.

LSU Message Count Controlled Link State Update Algorithm in QoS Routin (LSU 메시지 수를 제어 가능한 QoS 라우팅 링크 상태 갱신 알고리즘)

  • Cho, Kang-Hong;Kim, Nam-Hoon
    • Journal of the Korea Society of Computer and Information
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    • v.17 no.6
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    • pp.75-81
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    • 2012
  • This paper has proposed Message Count Control Mechanism based Link State Update(LSU) Algorithm that has not had a strong influence on the depreciation of QoS routing performance. Most existing LSU algorithms have the limit that cannot control the count of LSU message. Especially, adaptive algorithms have a bad performance when traffic are excessive and fickle. We classify as the importance of LSU message that have a influence on available bandwidth and determine the transmission of the message according to update rate per a unit of time. We have evaluated the performance of the proposed model and the existing algorithms on MCI simulation network using the performance metric as the QoS routing blocking rate and the mean update rate per link, it thus appears that we have verified the performance of this algorithm.