• 제목/요약/키워드: Data packet

검색결과 1,635건 처리시간 0.031초

An Efficient Data Transmission Scheme for Logistics Vehicles (물류 차량을 위한 효율적인 데이터 전송 방법)

  • Kim, Jong-Hyo;Yang, Jung-Min
    • IEMEK Journal of Embedded Systems and Applications
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    • 제13권2호
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    • pp.93-100
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    • 2018
  • In this paper, we present a novel scheme of data transmission for logistics vehicles connected with 3G mobile communication networks. The proposed method enhances the efficiency of data transmission by varying the packet transmission period according to the vehicle speed and by reducing the amount of transmitted data using a reduced packet structure. The main contribution is to present the experimental verification in which the proposed method is applied to commercial logistics vehicles that operate with networked data transmission modules. Being compared with the existing method, the proposed scheme shows superior performance in terms of data reduction and transmission speed.

A Method for the Performance Ehancement of PRMA Protocol for Mobile Voice/Data Integration (음성/데이터 통합형 PRMA 프로토콜의 성능 개선 기법)

  • 송재섭;김연수
    • The Journal of Korean Institute of Communications and Information Sciences
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    • 제25권3B호
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    • pp.423-430
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    • 2000
  • Future microcellular systems will require distributed network control. A packet-switched network is suitable for this requirement. The packet reservation multiple access(PRMA) is a Reservation-ALOHA like protocol for wireless terminals to transmit packet speech to a base station. It allows spatially distributed users in cellular systems to transmit packeted voice and data to a common base station using a shared channel. In the existing PRMA, the problem is that the voice packets may collide with the data packets due to simultaneous channel access. the problem may be a major performance degradation factor to a voice and data mixed system. We propose a new PRMA method that integrates voice and data traffic efficiently by resolving the collision problem between data and voice packets. The proposed PRMA method gives a performance improvement than the existing PRAMA method in terms of voice packet dropping probability and data delay characteristic. From analytic results, we can confirm that the proposed PRMA method show a performance improvement than the existing PRMA protocol.

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Optimization of the packet size to enhance the voice quality of the VOIP system (VOIP 음질 개선을 위한 패킷 크기의 최적화)

  • 임강빈;정기현;최경희
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • 제40권9호
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    • pp.373-383
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    • 2003
  • In this paper we discuss the effect of the delay limit and the packet size related to the quality of service on a VoIP system using the Internet. We also provide a guideline to determining the optimal packet size of the voice data for a given delay limit. Empirical studies are done with two personal computers connected through the packet switched public IP network. The sender encodes the voice signal from the microphone to get PCM and ADPCM data and sends the data to the receiver using UDP packets. The receiver plays the reconstructed voice from the stream with lost and delayed packets. The quality of the reconstructed voice is evaluated offline by the MNB (Measuring Normal Block) method using the data acquired from the both sides. The result shows that under the delay limit of 100ms for 40Kbps, 32Kbps and l6Kbps of ADPCM data, the minimum packet size should be 300bytes, 400bytes and 600bytes respectively and the maximum packet size should be l200bytes commonly for the best quality of voice.

A Dynamic Packet Recovery Mechanism for Realtime Service in Mobile Computing Environments

  • Park, Kwang-Roh;Oh, Yeun-Joo;Lim, Kyung-Shik;Cho, Kyoung-Rok
    • ETRI Journal
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    • 제25권5호
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    • pp.356-368
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    • 2003
  • This paper analyzes the characteristics of packet losses in mobile computing environments based on the Gilbert model and then describes a mechanism that can recover the lost audio packets using redundant data. Using information periodically reported by a receiver, the sender dynamically adjusts the amount and offset values of redundant data with the constraint of minimizing the bandwidth consumption of wireless links. Since mobile computing environments can be often characterized by frequent and consecutive packet losses, loss recovery mechanism need to deal efficiently with both random and consecutive packet losses. To achieve this, the suggested mechanism uses relatively large, discontinuous exponential offset values. That gives the same effect as using both the sequential and interleaving redundant information. To verify the effectiveness of the mechanism, we extended and implemented RTP/RTCP and applications. The experimental results show that our mechanism, with an exponential offset, achieves a remarkably low complete packet loss rate and adapts dynamically to the fluctuation of the packet loss pattern in mobile computing environments.

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Analysis of TCP packet by Protocol Analysis module Design (프로토콜 분석모듈 설계에 의한 TCP 패킷 분석)

  • Eom, Gum-Yong
    • Proceedings of the KIEE Conference
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    • 대한전기학회 2004년도 학술대회 논문집 정보 및 제어부문
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    • pp.234-236
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    • 2004
  • Transmission control protocol(TCP) is protocol used in internet. TCP is seldom transmission error and is protocol based on wire environment. TCP uses 3 way handshake ways, data transmission control through windows size, data transmission control through reception confirmation, sliding window for packet delivery. In this study, designed TCP packet ion module for analyze the TCP segments & correct information about TCP. TCP capture in internet using designed TCP module and analysed TCP segments composition. Through this, could analyze the correct information of protocol in network.

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Efficient Method for Exchanging Data between DDS Middlewares based on Adaptive Packet Transmission (적응형 패킷 전송에 기반한 DDS 미들웨어 간의 효율적인 데이터 교환 방법)

  • Ahn, Sung-Woo
    • Journal of the Korea Institute of Information and Communication Engineering
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    • 제16권6호
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    • pp.1229-1234
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    • 2012
  • In this paper, we analyze the problems that the DDS middleware, which is a standard data-centric communication interface, uses the fixed packet transmission method by the pre-defined protocol for exchanging data packets. The packet transmission method selected in a fixed manner cannot handle appropriately the increasing of resource overhead in an environment where the load of the DDS network changes dynamically. If the load on the node and network exceeds the threshold, the performance of the packet transmission may be degraded rapidly. This results in a failure of ensuring the real-time characteristic of DDS middleware. To solve this problem, we propose the scheme of the adaptive packet transmission for adjusting the transmission method in real-time based on the overhead on the DDS network.

모바일 오피스 서비스 지원을 위한 ADSRC 패킷 통신 시스템

  • Lee, Hyun;An, Dong-Hyun;Shin, Chang-Sub;Im, Chun-Sik;Park, Se-Ho;Cho, Kyung-Rok
    • Information and Communications Magazine
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    • 제19권9호
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    • pp.77-85
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    • 2002
  • In this paper, we introduce an ADSRC(hdvanced DSRC) OFDM packet communication system which has been developed by ETRI. The ADSRC system is targeted to provide high terminal mobility, high data rate and seamless service in roadside environment for mobile office services. We discuss the requirements of the ADSRC communication system for mobile office services, and the system design specification to meet them with regard to air interface. The ADSRC packet communication systems consist of the MAC processor block, the OFDM packet modem block and the RF block. The MAC processor block handles medium access control and the test. The OFDM packet modem transmits data packets at up to 24Mbps adaptively and recovers the data from RF block. We describe the ADSRC packet communication system architecture and the ADSRC system protocol.

Implementation and evaluation of lost packet recovery using low-bitrate redundant audio data (저비트율 잉여오디오 정보를 이용한 손실 패킷 복구 방법의 구현 및 성능 평가)

  • 박준석;고대식
    • Journal of the Korean Institute of Telematics and Electronics S
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    • 제35S권7호
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    • pp.1-5
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    • 1998
  • In this paper, recovery method with high-bitrate and low-bitrate coder was implemented in order to recover consecutive packet loss over the Internet. LPC was used as redundant audio data for recover of lost packets and RTP parcket format was modified for accommodation of redundant data. In measuring results using random packet loss rate with three redundant datra in every packet, it has shown that recovery rate was 80% in los rate of 50%. Since the processing delay for recovery of the lost packet was 200ms, this recovery method can be applied to real-time Internet sevice such as Internet phone.

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A Model to Investigate the Security Challenges and Vulnerabilities of Cloud Computing Services in Wireless Networks

  • Desta Dana Data
    • International Journal of Computer Science & Network Security
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    • 제23권10호
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    • pp.107-114
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    • 2023
  • The study provides the identification of vulnerabilities in the security issues by Wireless Network. To achieve it the research focus on packet flow analysis, end to end data communication, and the security challenges (Cybercrime, insider threat, attackers, hactivist, malware and Ransomware). To solve this I have used the systematic literature review mechanisms and demonstrative tool namely Wireshark network analyzer. The practical demonstration identifies the packet flow, packet length time, data flow statistics, end- to- end packet flow, reached and lost packets in the network and input/output packet statics graphs. Then, I have developed the proposed model that used to secure the Wireless network solution and prevention vulnerabilities of the network security challenges. And applying the model that used to investigate the security challenges and vulnerabilities of cloud computing services is used to fulfill the network security goals in Wireless network. Finally the research provides the model that investigate the security challenges and vulnerabilities of cloud computing services in wireless networks

Epicenter Estimation Using Real-Time Event Packet of Quanterra digitizer (Quanterra 기록계의 실시간 이벤트 패킷을 이용한 진앙 추정)

  • Lim, In-Seub;Sheen, Dong-Hoon;Shin, Jin-Soo;Jung, Soon-Key
    • Geophysics and Geophysical Exploration
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    • 제12권4호
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    • pp.316-327
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    • 2009
  • A standard for national seismological observatory was proposed on 1999. Since then, Quanterra digitizer has been installed and is operating on almost all of seismic stations which belong to major seismic monitoring organizations. Quanterra digitizer produce and transmit real-time event packet and data packet. Characteristics of event packet and arrival time of each channel's data packet on data center were investigated. Packet selection criteria using signal to noise ratio (hereafter SNR) and signal period from real-time event packet based on 100 samples per second (hereafter sps) velocity data were developed. Estimation of epicenter using time information of the selected event packet were performed and tested. A series of experiment show that event packets were received approximately 3~4 second earlier than data packets and the number of event packet was only 0.3% compare to data packets. Just about 5% against all of event packets were selected as event packet were related P wave of real earthquake. Using the selected event packets we can estimate an epicenter with misfit less than 10 km within 20 sec for local earthquake over magnitude 2.5.