• Journal of the Institute of Electronics Engineers of Korea SD
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• v.42 no.2 s.332
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• pp.89-96
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• 2005

In this paper, we propose a customer premise platform for processing multimedia data service on the ATM network. The proposed platform has a specific AAL2 processor that includes AAL2 protocol and scheduler algorithm so as to off-load large potion of burden from host processor and make it easy to process multimedia data from the ATM network in real time compared with conventional platform in which AAL/ATM tasks are processed by software. The ATS scheduler that is implemented based on 2-level time slot ring provides a simple and efficient method for scheduling data of VBR-rt, UBR and CBR traffics. TMS320C5402 DSP is used to process voice-related tasks such as voice compression and voice packet manupulation and AAL2 processor is implemented on $0.35\;{\mu}m$ process line. We implemented the customer premise equipment for VoDSL service and tested the proposed platform on a test bed network. The experimental results show that the proposed equipment has the call success rate of $97\%$ at least and provides voice service of toll-qualify.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.129-132
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• 2003

In this paper, which sees the Strong-ARM SA1110 it used the main CPU and RTP in VoIP system. It will be able to apply the information communication field it embodied. It used the Tynux_box2 with the hardware side and it composed a VOIP system. And it used the RTP which is a real-time protocol in software control portion. The development environment of the paper that used the Target board and a Linux PC for connection used the RS-232C, USB connection, Ethernet LAN. The VoIP the environment for a communication used the wave file in the substitution which changes analog signal with the digital signal. And For the communication of the both sides it used the socket. This paper explained the fact that against a general technique from the operation of VoIP system. Using the Embedded linux development board which explained an operational process of the RTP protocol.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.1
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• pp.179-189
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• 2000

Recently, due to the increasing interests in deep sea development, all possible efforts to the development of underwater unmanned working vehicles such as AUV(Autonomous Underwater Vehicle) or underwater robot are exerted. This paper proposes a new efficient acoustic-based underwater image data communication system, which ensures a certain level of maximum throughput regardless of the propagation delay of ultrasonic and allowsfast data transmission through the multiple ultrasonic communication channel. Proposed system consists of an acoustic transducer which operates at 136kHz center frequency and it's 10kHz bandwidth, pre-amplifier, $\pi/4 QPSK$(Quadrature Phase Shift Keying) modulation/demodu-lation method, image compressing method using JPEG technique and modified Stop & Wait protocol. The experimental result of the system make it possible to transfer the underwater image as a high throughput at the basin test. The results of test are also verified which allows to desirable transmission performance compared with the existing developed system and the possibility to put the practical use of survey and investigation in the water.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39C no.5
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• pp.435-446
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• 2014

Even if it is important to transmit high rate multimedia information without any transmission errors for surveillance systems, it is difficult to achieve error-free transmission due to infra-less adhoc networks. In order to reduce the transmission errors furthermore, additional signal overheads or retransmission of signals should be required, but they may lead to transmission delay. This paper represents a study on low delay and adaptive video transmission for the unmanned surveillance systems by developing system protocols. In addition, we introduce an efficient and adaptive control algorithm using system parameters for exploiting unmanned surveillance system properly over multi-channels.

• The KIPS Transactions:PartC
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• v.16C no.2
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• pp.237-244
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• 2009

The congestion in wireless sensor network increases the ratio of data loss and causes the delay of data. The existing congestion protocols for wireless sensor network reduces the amount of transmission by control the sampling frequency of the sensor nodes related to the congestion when the congestion has occurred and was detected. However, the control method of sampling frequency is not applicable on the situation which is sensitive to the temporal data loss. In the paper, we propose a new congestion control, ACT - Adaptive Congestion conTrol. The ACT monitors the network traffic with the queue usage and detects the congestion based on the multi level threshold of queue usage. Given network congestion, the ACT increases the efficiency of network by adaptive flow control method which adjusts the frequency of packet transmission and guarantees the fairness of packet transmission between nodes. Furthermore, ACT increases the quality of data by using the variable compression method. Through experiment, we show that ACT increases the network efficiency and guarantees the fairness to sensor nodes compared with existing method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.8
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• pp.2090-2095
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• 1998

In this paper, we measured voice packet loss over the Korean Internet and analyzed percentage articulation by variation of the packet loss. To do this, we reviewed real-time transmission service based on RTP/UDP/IP and test method of the transmission quality. and implemented the real-time speech transmission system using GSM and UDP/IP. Monosyllable list has been chosen for the percentage articulation test, each voice packet has been coded and compressed by GSM and it has sequence number to measured packet loss and to recover out-of-order packets. In transmission results using seven router over the Korean Internet, we have show that loss rates reached 1.6% (unload), 22.5%(load) and loss rates after packet recovery by resequencing and FEC are from 9% to 35%. Finally, we have shown that percentage articulations by variation of the network traffic are Table 4.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.394-400
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• 2002

In digital speech communications, the quality of service can be increased by speech coding scheme that is adaptive to the error rate of voice packet transmission. However, current communication protocol in cellular and internet communications does not provide the function that transmits the channel error information. To solute this problem, in this paper, new method for real-time transmission of channel error information is proposed, where channel error information is embedded in voice packet. The proposed method utilizes the pulse positions of codevector in ACELP speech codec, which results in little degradation in speech quality and low false alarm rate. The simulations with various speech data show that the proposed method meets the requirement in speech quality, detection rate, and false alarm rate.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.2
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• pp.264-270
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• 2006

In this paper we verified that the window based TCP performance of throughput can be improved by the traffic connection efficiency. and have studied the performance of traffic congestion control that is controlling transmission rate. In wireless network, the bidirectional node is run by estimating the usage rate of link of error control idle and the throughput is shown by transmitting segments. The throughput rate shows almost no delay due to the bidirectional traffic connection efficiency up to the allowable point as increasing the transport rate by the critical value, depending on the size of end-to-end node queue of the increase of transport rate. This paper reports the performance improvement as the number of feedback connection traffic congestion control increases because of the increase of the number of asynchronous transport TCP connections.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.2 no.3
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• pp.417-429
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• 1998

The advance of the MPEG (Moving Picture Expert Group) and the DSP (Digital Signal Processing) technologies lead the emergence of the information appliances, which are gradually digitalized and embedded the high-speed networking function. However, there are some difficulties in establishing any one specific model, since standardization of the scale and function of home ATM network is being progressed by standardization organizations and no absolute model has been defined. This paper consider topologies for a home ATM network, such a star-type, tree-type, and mesh-type topology, by comparing the structure, functional characteristics and performance. From this analysis we suggest the design method of the home ATM network.

• Proceedings of the Korean Information Science Society Conference
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• 2005.07a
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• pp.313-315
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• 2005

사회적으로 큰 관심의 대상이 되고 있는 무선 인터넷은 유선 인터넷과 달리 기술 환경과 그 특성상 여러 가지 제약점들을 가지고 있다. 대역폭이 낮고, 접속이 빈번하게 끊기며, 단말기내의 컴퓨팅 파워가 낮고 화면이 작다. 또한 사용자의 이동성 문제와 네트워크 프로토콜, 보안 등에서 아직 기술적으로 부족한 부분을 보이고 있다. 그리고 급속도로 증가하는 수요에 따라 무선 인터넷 서버는 대용량 트래픽을 처리할 수 있는 확장성이 요구되어지고 있다. 이에 본 논문에서는 무선 인터넷 프록시 서버 클러스터를 사용하여 앞에서 언급된 무선 인터넷의 문제와 요구들을 캐싱(Caching), 압축(Distillation) 및 플러스터(Clustering)를 통하여 해결하려고 한다. TranSend는 클러스터링 기반의 무선 인터넷 프록시 서버로 제안된 것이나 시스템적인(Systematic) 방법으로 확장성을 보장하지 못하고 불필요한 모듈간의 통신구조로 인해 복잡하다는 단점을 가진다. 기존 연구에서 시스템적인 방법으로 확장성을 보장하는 CD-A라는 구조를 제안하였으나 이 역시 모듈간의 불필요한 통신 구조를 가진다는 단점을 있다. 이에 본 논문에서는 시스템적인 확장성과 단순한 구조를 가지는 새로운 클러스터링 기반의 무선 인터넷 프록시 서버를 제안한다. 16대의 컴퓨터를 사용하여 실험을 수행하였고 실험 결과 TranSend 시스템과 CD-A 시스템에 비해 각각 $216\%,\;40\%$의 성능 향상을 보였다.