• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.10
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• pp.1-9
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• 2004

User-Perceived application-level performance is a key to the adoption and success of CDMA 2000. To predict this performance in advance, a detailed end-to-end simulation model of a CDMA network was built to include application traffic characteristics, network architecture, network element details, and protocol features. We assess the user application performance when a Radio Access Network (RAN) and a Core Network (CN) adopt different transport architectures such as ATM and If. For voice Performance, we found that the vocoder bypass scenario shows 8% performance improvement over the others. For data packet performance, we found that HTTP v.1.1 shows better performance than that of HTTP v.1.0 due to the pipelining and TCP persistent connection. We also found that If transport technology is better solution for higher FER environment since the IP packet overhead is smaller than that of ATM for web browsing data traffic, while it shows opposite effect to small size voice packet in RAN architecture. Though simulation results we showed that the 3G-lX EV system gives much better packet delay performance than 3G-lX RTT, the main conclusion is that end-to-end application-level performance is affected by various elements and layers of the network and thus it must be considered in all phases of the technology evolution process.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.6A
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• pp.558-564
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• 2007

3G Long Term Evolution, which aims for various mobile multimedia service provision by enhanced wireless interface, proposes VoIP-based voice service through a Packet Switching (PS) domain. As delay and loss-sensitive VoIP traffic flows through the PS domain, more challenging technical difficulties are expected than in Circuit Switching (CS) domain based VoIP services. Moreover, since 3G LTE, which adopts the OFDM as its physical layer, introduces Physical Resource Block (PRB) as a unit for transmission resources, new types of resource management schemes are needed. This paper proposes a PRB scheduling algorithm of MAC layer for VoIP service in 3G LTE and shows the simulation results. The proposed algorithm has two key parts; dynamic activation of VoIP priority mode to satisfy VoIP QoS requirements and adaptive adjustment of the priority mode duration in order to minimize the degradation of resource utilization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.8A
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• pp.777-784
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• 2010

Recently, the necessity of underwater communication and demand for transmitting and receiving various data such as voice or high resolution image data are increasing as well. The performance of underwater acoustic communication system is influenced by characteristics of the underwater communication channels. Especially, ISI(inter symbol interference) occurs because of delay spread according to multi-path and communication performance is degraded. In this paper, we study the OFDM technique to overcome the delay spread in underwater channel and by using CP, we compensate for delay spread. But PAPR which OFDM system has problem is very high. Therefore, we use DFT-spread OFDM method to avoid nonlinear distortion by high PAPR and to improve efficiency of amplifier. DFT-spread OFDM technique obtains high PAPR reduction effect because of each parallel data loads to all subcarrier by DFT spread processing before IFFT. In this paper, we show performance about delay spread through OFDM system and verify method that DFT spread OFDM is more suitable than OFDM for underwater communication. And we analyze performance according to two subcarrier mapping methods(Interleaved, Localized). Through the simulation results, performance of DFT spread OFDM is better about 5~6dB at $10^{-4}$ than OFDM. When compared to BER according to subcarrier mapping, Interleaved method is better about 3.5dB at $10^{-4}$ than Localized method.

• The Transactions of the Korea Information Processing Society
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• v.5 no.12
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• pp.3047-3062
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• 1998

This paper deseribes the desihn and the implementation of the Realtime Transport Protocol(RTP)/ Rdaltime Control Protocol(RTCP) (RFC 1889,1890) that is used to transmit the audio/video data to any destination and to feedback the Quality of Service (QoS) information of the received media data to the sender, in the teleconferencing systems proposed by ITU-T. These protocols are implemented with multi thead technique and run on top of UDP/IP-Multicast through the socket interface as the underlying protocol. The upper layer is impelmented such that in can be accessed by the H245 comference control protocol. The RTP packetizes the digitized audio/video data from the encoder info a fixed format, and multieast to the participants. The RTCP monitors RTP packets and extracts the QoS values from it such as round-trip delay, jiter and packet loss to form RTCP packets and non periokically sends them to the sender site. In this Paper, we also descritx the study of measurement and analysis for QoS factors that observed on performing teleconferencing system over Internet. The results from this experiment is indicate that RTT and Jitter value are acceptable even entwork load is high. However, it appears that packet loss rate is high in daytime and most losses periods have length one or two.

• The Journal of the Acoustical Society of Korea
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• v.20 no.6
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• pp.34-39
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• 2001

An adaptive multi-rate (AMR) speech coder was adopted as a standard of W-CDMA by 3GPP and ETSI. The AMR coder is based on the CELP algorithm operating at rates ranging from 12.2 kbps down to 4.75 kbps, and it is a source controlled codec according to the channel error conditions and the traffic loading. In this paper, we implement the DSP S/W of the AMR coder using OakDSPCore. The implementation is based on the CSD17C00A chip developed by C&S Technology, and it is tested using test vectors, for the AMR speech codec, provided by ETSI for the bit exact implementation. The DSP B/W requires 20.6 MIPS for the encoder and 2.7 MIPS for the decoder. Memories required by the Am coder were 21.97 kwords, 6.64 kwords and 15.1 kwords for code, data sections and data ROM, respectively. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

• Journal of KIISE:Computing Practices and Letters
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• v.14 no.1
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• pp.101-105
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• 2008

This paper proposes an analytical model to calculate VoIP (Voice of IP) capacity over wireless LANs with frequent bit errors. Since the traditional analytical models for VoIP capacity have not included the effect of bit errors, simulations ould only evaluate VoIP capacity over erroneous channels. For analytically accurate estimation of VoIP capacity over noisy channels, we extend the conventional model to include the effect of propagation errors, end-to-end delay, voice quality, the waiting time in AP(Access Point). The experiments show that our model predicts the VoIP capacity of a given network within the range from 3% to 9% difference comparing with the simulation results.

• Journal of the Korea Society of Computer and Information
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• v.6 no.1
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• pp.45-50
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• 2001

Congestion may take place in the ATM network because of high-speed cell transmission features, and cell delay and loss also can be caused by unexpected traffic variation. Thus. traffic control mechanisms are needed. One of them to decrease congestion is the cell shaping. This paper proposes a hybrid type cell shaper composed of a Leaky Bucket with token pool, EWMA with time window, and a spacing control buffer. The simulator BONeS with the ON/OFF traffic source model evaluates the performance of the proposed cell shaping method. Simulation results show that the cell shaping concerning the respective source traffics is adapted to and then controlled on the mean bit rate.

• Proceedings of the Korea Contents Association Conference
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• 2003.11a
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• pp.347-351
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• 2003

It is important to consider not only audio/video but presentation way, in video conferencing, seminar, lecture based network communication. The method to transmit image of a chalkboard used in traditional seminar, conference, lecture are one of difficulties for remote environment since it requires techniques high-resolution transmission and talking films. In this paper we implemented real-time multimedia shared board system for shared effective multimedia presentation using connection-oriented socket of TCP. In this system, decreasing cost of related system construction, improve of data reliableness and interaction, presentation delay time be able to minimize.

• Journal of the Korea Society of Computer and Information
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• v.18 no.2
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• pp.59-68
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• 2013

Efficient resource management and hand-over schemes are necessary to maintain consistent QoS because it may be severely defected by some delay and information loss during hand-over in LTE-Advanced networks. This paper proposes a resource management scheme based on the load control to support consistent QoS for heterogeneous services during hand-over in OFDMA based systems. Various multimedia services with different requirements for resource are able to be serviced simultaneously because service continuity can be provided by our proposed scheme. Simulation results show that it provides better performances than the conventional one with the measure of hand-over failure rate and packet loss rate.

• Proceedings of the Korea Information Processing Society Conference
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• 2006.11a
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• pp.617-620
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• 2006

개방화시대에 제품의 국제경쟁력을 확보하기 위해서는 Unix나 Tandem 환경의 시스템운영관리 측면에서 실시간 모니터링 운영의 효율성이 증대 하고 있고 이를 바탕으로 서비스의 품질 및 기술력 향상과 고객만족 서비스 등 경제성이 필수적으로 요구되고 있다. 기존의 시스템관리 기능의 운영측면에서 보다는 보다 더 다양한 고객의 요구가 반영이 가능한 새로운 형태의 실시간 업무지원이 가능한 모니터링 시스템 개발이 필요한 시점이다. 기존의 시스템관리 기능에서 보다 업무지원이 기능이 강하된 모니터링 툴 이다. 즉 업무(Application) 프로세스의 장애현상에 따른 무응답을 사전 감시하고 System의 자원사용현황을 개발 및 운영자 위주의 실시간으로 감시환경 모니터링 서비스 제공이 가능한 기능으로 설계 따라서 시스템관리에서 보다 안정적인 운영을 제공 할 수 있다. 본 연구에서는 업무 프로세스의 장애를 점검할 수 있는 모니터링 툴 검토하여 실무자를 통하여 업무장애 유형에 따른 감시항목 및 의견 수렴, 요건정의에 따른 업무설명회 수행, 탠덤 시스템 모니터링 프로젝트 수행 필요한 기능을 개발 할 것이다. 또한 시스템 운영자 입장에서 실시간 모니터링 서비스를 통하여 업무(Application)프로그램의 전송지연 및 장애 원인을 한눈에 파악 점검 하도록 하며 각 기능별 시스템(System)별 장애 감시 통하여 사용자에서 현재 상태를 종합음성경보(SMS) 구축 연계하여 지원한다. 따라서, 본 논문에서는 중요한 요소로 기존의 많은 Unix나 Tandem 환경의 시스템관리 기능을 비교 분석을 통하여 보다 효율성이 높은 고객서비스 기능과 업무지원이 가능한 실시간 시스템관리 모니터링 툴을 개발 한다.