• Title/Summary/Keyword: packet service system

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The implementation of the Remote Control and Measurement Systems using CDMA Modem (CDMA 모뎀을 이용한 원격 제어 및 계측 시스템 구현)

  • Lee, Myung-Eui
    • Journal of Advanced Navigation Technology
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    • v.16 no.2
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    • pp.351-359
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    • 2012
  • This paper deals with the design and development of the remote control and measurement systems using CDMA(Code Division Multiple Access) data modem. We propose a bi-directional data communication link without the public IP address in CDMA modem device by the TCP/IP packet and SMS(Short Message Service) communication. The remote control and measurement systems are implemented by a Telit WM-800 modem as DCE(Data Communication Equipment), and Atmel AT89C51 microcontroller as DTE(Data Terminal Equipment). The user application software for the control and measurement system user, and the firmware software of device drivers for peripherals such as a digital input/output device, AD/DAC(Analog to Digital/Digital to Analog Converter), LCD, and temperature/humidity sensor are written in Microsoft C and Keil C language respectively for further various applications. The experimental result of the proposed control and measurement systems implemented in this paper is evaluated via real-time experiments, which works well as designed.

Implementation of π/4-DQPSK Modem for Maritime Digital Communication in VHF Band (VHF 대역 해상 디지털 통신용 π/4-DQPSK 모뎀 구현)

  • Kwak, Jaemin
    • Journal of Advanced Navigation Technology
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    • v.18 no.6
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    • pp.541-545
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    • 2014
  • Rec. ITU-R M.1842-1 is international recommendation for VHF band communication guideline in maritime mobile service RR Appendix 18 channels. In this paper, we simulate 28.8 kbps VHF ${\pi}$/4-DQPSK digital baseband modem compatible with the recommendation, then it is designed and implemented with FPGA. Cazac sequence is used as a preamble since packet format is not defined untill now in the recommendation. Baseband modem is designed by VHDL language and implemented on NEXYS4 development platform having Atrix7 FPGA chip from Xilinx. For wireless communication test of total prototype system, ADC/DAC board is implemented and EV9730 RF module is utilized. From the experimental results, implemented FPGA modem shows spectral bandwidth of 25 kHz and successful data exchanges between tx and rx communication platform.

A Scheme for Push/Pull Buffer Management in the Multimedia Communication Environments (멀티미디어 통신 환경에서 Push/Pull 버퍼 관리 기법)

  • Jeong, Chan-Gyun;Lee, Seung-Ryong
    • The Transactions of the Korea Information Processing Society
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    • v.7 no.2S
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    • pp.721-732
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    • 2000
  • Multimedia communication systems require not only high-performance computer hardwares and high-speed networks, but also a buffer management mechanism to process many data efficiently. Two buffer handling methods, Push and Pull, are commonly used. In the Push method, a server controls the flow of dat to a client, while in the Pull method, a client controls the flow of data from a server. Those buffering schemes can be applied to the data transfer between the packet receiving buffer, which receives media data from a network server, and media playout devices, which play the recived media data. However, the buffer management mechanism in client-sides mainly support either one of the Push or the Pull method. Consequently, they have some limitations to support various media playout devices. Futhermore, even though some of them support both methods, it is difficult to use since they can't provide a unified structure. To resolved these problems, in this paper, we propose an efficient and flexible Push/Pull buffer management mechanism at client-side. The proposed buffer management scheme supports both Push and Pull method to provide various media playout devices and to support buffering function to absorb network jitter. The proposed scheme can support the various media playback devices using a single buffer space which in consequence, saves memory space compared to the case that a client keeps tow types of buffers. Moreover, it facilitates the single buffer as a mechanism for the absorbing network jitter effectively and efficiently. The proposed scheme has been implemented in an existing multimedia communication system, so called ISSA (Integrated Streaming Service Architecture), and it shows a good performance result compared to the conventional buffering methods in multimedia communication environments.

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Smart Grid Cooperative Communication with Smart Relay

  • Ahmed, Mohammad Helal Uddin;Alam, Md. Golam Rabiul;Kamal, Rossi;Hong, Choong Seon;Lee, Sungwon
    • Journal of Communications and Networks
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    • v.14 no.6
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    • pp.640-652
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    • 2012
  • Many studies have investigated the smart grid architecture and communication models in the past few years. However, the communication model and architecture for a smart grid still remain unclear. Today's electric power distribution is very complex and maladapted because of the lack of efficient and cost-effective energy generation, distribution, and consumption management systems. A wireless smart grid communication system can play an important role in achieving these goals. In this paper, we describe a smart grid communication architecture in which we merge customers and distributors into a single domain. In the proposed architecture, all the home area networks, neighborhood area networks, and local electrical equipment form a local wireless mesh network (LWMN). Each device or meter can act as a source, router, or relay. The data generated in any node (device/meter) reaches the data collector via other nodes. The data collector transmits this data via the access point of a wide area network (WAN). Finally, data is transferred to the service provider or to the control center of the smart grid. We propose a wireless cooperative communication model for the LWMN.We deploy a limited number of smart relays to improve the performance of the network. A novel relay selection mechanism is also proposed to reduce the relay selection overhead. Simulation results show that our cooperative smart grid (coopSG) communication model improves the end-to-end packet delivery latency, throughput, and energy efficiency over both the Wang et al. and Niyato et al. models.

Simulation Analysis of User Grouping Algorithms for Massive Smart TV Services (시뮬레이션을 이용한 대규모 스마트 TV 서비스 제공을 위한 사용자 그룹핑 알고리즘 성능 분석)

  • Jeon, Cheol;Lee, Kwan-Seob;Jou, Wou-Seok;Jeong, Tai-Kyeong Ted.;Han, Seung-Chul
    • Journal of the Korea Society for Simulation
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    • v.20 no.1
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    • pp.61-67
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    • 2011
  • Smart TV System will lead to drastic change of communication and media industries as one of the emerging next generation network services. However, when the number of concurrent users increases rapidly, the issue of service quality degradation occurs because providing services to many users simultaneously stresses both the server and the network. The server limitation can be circumvented by deploying server clusters. but the network limitation is far less easy to cope with, due to the difficulty in determining the cause and location of congestion and in provisioning extra resources. In order to alleviate these problems, a number of schemes have been developed. Prior works mostly focus on reducing user-centric performance metrics of individual connection, such as the round-trip time(RTT), downloading time or packet loss rate, but tend to ignore the network loads caused by the concurrent connections or global network load balance. In this work, we make an in-depth investigation on the issue of user grouping for massive Smart TV services through simulations on actual Internet test-bed, PlanetLab.

VoIP Quality Metric and Quality-based Accounting Scheme (VoIP 품질 측량 도구 및 품질 기반의 요금 부과 방안 연구)

  • Jung, Youn-Chan;Ann, Ibanez Al
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.35 no.1B
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    • pp.27-34
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    • 2010
  • As VoIP systems move to wireless environments with much higher average packet loss rates than wired networks, it becomes less possible for the network to assure a reasonable QoS. So, real-time quality monitoring for mobile VoIP applications is an important issue to be explored. In this paper, we explore perceptual quality dependency on two parameters: the burst loss rate and average burst length. Also, we propose a simple 'moving average' approach with $\alpha$ aiming to measure those parameters on real-time basis. In order to find how accurately the two parameters measured estimate the real perceptual quality, we compare actual measured PESQ scores with estimated value by matching the measured quality metric to the trained MOS table. Finally, we propose the quality-based accounting system, which can set obvious continuities between quality and billing.

Implementation of QoS-Measuring System for Voice over IP (VoIP(Voice over Internet Protocol) 품질 측정을 위한 UA(User Agent) 및 서버 기능 연구)

  • Kang, Hyun-Joong;Nam, Heung-Woo
    • Journal of the Korea Society of Computer and Information
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    • v.12 no.1 s.45
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    • pp.137-144
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    • 2007
  • Advances in networking technology digital media, and codecs have made it possible for the Internet evolves into a Broadband convergence Network (BcN) and provides various services including Voice over Internet Protocol (VoIP) and IPTV over their high-speed IP networks. In order for the Internet to make a profit as traditional Public Switched Telephone Network (PSTN), it must provide high qualify VoIP services. Therefore, real time qualify measurement framework is the most important requisite to provide VoIP service. For this, IETF (Internet Engineering Task Force) defined RTCP-Extended Reports (RTCP-XR) that extend RTCP (Real-Time Transport Protocol Control Protocol). However, procedure and method tot actually VoIP qualify measurement did not recommended nothing but defined item to measure voice quality. Our objective in this paper is to describes a practical measuring framework for end-to-end QoS of switched voice packet in an IP environment. It includes concepts as well as step-by-step procedures for measuring packetized voice streams. It also proposes new formats that extend RTCP-XR's concept.

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Match Field based Algorithm Selection Approach in Hybrid SDN and PCE Based Optical Networks

  • Selvaraj, P.;Nagarajan, V.
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.12 no.12
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    • pp.5723-5743
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    • 2018
  • The evolving internet-based services demand high-speed data transmission in conjunction with scalability. The next generation optical network has to exploit artificial intelligence and cognitive techniques to cope with the emerging requirements. This work proposes a novel way to solve the dynamic provisioning problem in optical network. The provisioning in optical network involves the computation of routes and the reservation of wavelenghs (Routing and Wavelength assignment-RWA). This is an extensively studied multi-objective optimization problem and its complexity is known to be NP-Complete. As the exact algorithms incurs more running time, the heuristic based approaches have been widely preferred to solve this problem. Recently the software-defined networking has impacted the way the optical pipes are configured and monitored. This work proposes the dynamic selection of path computation algorithms in response to the changing service requirements and network scenarios. A software-defined controller mechanism with a novel packet matching feature was proposed to dynamically match the traffic demands with the appropriate algorithm. A software-defined controller with Path Computation Element-PCE was created in the ONOS tool. A simulation study was performed with the case study of dynamic path establishment in ONOS-Open Network Operating System based software defined controller environment. A java based NOX controller was configured with a parent path computation element. The child path computation elements were configured with different path computation algorithms under the control of the parent path computation element. The use case of dynamic bulk path creation was considered. The algorithm selection method is compared with the existing single algorithm based method and the results are analyzed.

Advanced ICMP Traceback Mechanism Against DDoS Attack in Router (DDoS 공격에 대한 개선된 라우터 기반 ICMP Traceback iT법)

  • 이형우
    • Journal of the Korea Institute of Information Security & Cryptology
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    • v.13 no.6
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    • pp.173-186
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    • 2003
  • Distributed Denial-of-Service(DDoS) attack prevent users from accessing services on the target network by spoofing its origin source address with a large volume of traffic. The objective of IP Traceback is to determine the real attack sources, as well as the full path taken by the attack packets. Existing IP Traceback methods can be categorized as proactive or reactive dating. Proactive tracing(such as packet marking and messaging) prepares information for tracing when packets are in transit. Reactive tracing starts tracing after an attack is detected. In this paper, we propose a 'advanced ICW Traceback' mechanism, which is based on the modified pushback system with secure router mechanism. Proposed mechanism can detect and control DDoS traffic on router and can generate ICMP Traceback message for reconstructing origin attack source, by which we can diminish network overload and enhance Traceback performance.

Performance Analysis of Sensor Network Real-Time Traffic for Factory Automation in Intranet Environment (인트라넷 환경에서의 공장자동화를 위한 센서 망 실시간 트래픽 성능 평가)

  • Song, Myoung-Gyu;Choo, Young-Yeol
    • Journal of Korea Multimedia Society
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    • v.11 no.7
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    • pp.1007-1015
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    • 2008
  • In order to provide real-time data from sensors and instruments at manufacturing processes on web, we proposed a communication service model based on XML(eXtensible Markup Language). HTML(Hyper Text Markup Language) is inadequate for describing real-time data from manufacturing plants while it is suitable for display of non-real-time multimedia data on web. For applying XML-based web service of process data in Intranet environment, real-time performance of communication services was evaluated to provide the system design criteria. XML schema for the data presentation was proposed and its communication performance was evaluated by simulation in terms of transmission delay due to increased message length and processing delay for transformation of raw data into defined format. For transformation of raw data into XML format, we proposed two structures: one is the scheme where transformation is done at an SCC(Supervisory Control Computer) after receiving real-time data from instruments. the other is the scheme where transformation is carried out at instruments before the data are transmitted to the SCC. Performances of two structures were evaluated on a testbed under various conditions such as six packet sizes and offered loads of 20%, 50% and 80%, respectively. Test results show that proposed schemes are applicable to the systems in Ethernet 100BaseT network if total message traffic is less than 7 Mbps.

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