• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.25-30
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• 2005

We propose and design a fully conversed Ethernet and TDM transport system to guarantee qualify of service for Ethernet packet. Developed convergence system can support not only L2 VPN service and premium multimedia service based on MPLS protocol but also TDM leased line service, simultaneously. Developed convergence system has the advantage of providing high reliability that realizes high-quality and real time communications due to assign QoS profile and guarantee bandwidth per channel consist of VCGs without affect adjacent channels. Evaluation for proposed system was successfully performed within the ling network.

• Journal of KIISE:Information Networking
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• v.33 no.1
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• pp.91-102
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• 2006

We analyze the performance of MAC-layer Reed-Solomon error recovery in the cdma2000 1xEV-DO Broadcast and Multicast Services (BCMCSs) environment, with respect to the size of the ECB (Error Control Block) and the air-channel condition, and establish the relationship between ECB size, error recovery capacity and service latency. From this we propose an adaptive error recovery scheme which adjusts the size of the ECB to reflect the environment of the mobile nodes so as to meet the required service quality (target bit error-rate), while reducing the latency of real-time applications. Extensive simulation results show the effectiveness of our approach compared to the current static scheme. Proposed adaptive schemes achieves near optimal solution with respect to service latency while satisfying the required service quality.

• Aerospace Engineering and Technology
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• v.9 no.1
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• pp.67-71
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• 2010

In this study, an algorithm for blind modulation detection in EDGE systems is presented. EDGE introduces an 8PSK modulation to provide high-speed data rates in addition to the existing GSM system. A transmitter may switch dynamically the modulation and coding schemes for transmission of data according to the channel quality. To decode the data correctly, the receiver has to detect using only training sequence which modulation is being used. Basically the property of one radio block composed of four bursts to detect effectively the modulation scheme even under severe conditions is used. More specifically, the reference value calculated for received burst is accumulated with previous reference value to minimize statistically the false detection probability in one radio block. Also each burst data having different modulation from the modulation of the fourth burst is set to zero to improve the decoding performance because the reference of the fourth burst has the highest reliability.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.9
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• pp.1646-1653
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• 2008

In this paper, we propose multicarrier CDMA system using the concept of the modulation level-controlled system in order to make the system robust to the frequency selectivity and to provide a maximum data rate maintaining an acceptable transmission quality over various channel environments. This system selects higher data rate when the channel experiences the low delay spread and slow fading. On the other hand, when the fading changes very fast and the delay spread is very long, the system selects lower data rate. In both cases, the system controls the number of serial-to-parallel converted streams and the number of fed streams. This system has the fixed number of sub-carriers. So the product of tile number of serial-to-parallel converted streams and the number of fed streams is always kept constant. With the same data fed at different sub-carriers, the frequency diversity is achieved. And a RAKE receiver also is utilized to achieve path (time) diversity.

• The Journal of the Korea Contents Association
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• v.10 no.3
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• pp.80-91
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• 2010

Due to the convergence of broadcasting and communications, IPTV services are spotlighted as the that next-generation multimedia services. IPTV services should have functionality such as unlimited channel capacity, extension of media, QoS awareness and are required increasing traffic and quality control technology to adapt the attributes of IPTV service. Consequently, flow based quality control techniques are needed. Therefore, many studies for providing Internet QoS are performed at IETF (Internet Engineering Task Force). As the buffer management mechanism among IP QoS methods, active queue management method such as RED(Random Early Detection) and modified RED algorithms have proposed. However, these algorithms have difficulties to satisfy the requirements of various Internet user QoS. Therefore, in this paper we propose the Flow based AQM(Active Queue Management) algorithm for the multimedia services that request various QoS requirements. The proposed algorithm can converge the packet loss ratio to the target packet loss ratio of required QoS requirements. And we present a performance evaluation by the simulations using the ns-2.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.4
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• pp.891-898
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• 2014

The DVC (Distributed Video Coding) provides a theoretical basis for the implementation of light video encoder. Conventionally, lots of studies have been focused on the codec scheme of Stanford University that has a feedback channel to control the bit rate finely. However, the codec scheme can not evaluate the qualities of the frames reconstructed by the received parity bits at the decoder side. This paper presents an efficient method of estimating distortion by correcting the virtual channel noises in side information and then facilitating the measurements of the visual qualities. Through several simulations, it is shown that the proposed method is very efficient in estimating the visual qualities of the reconstructed WZ frames.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.11
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• pp.2405-2410
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• 2012

The 4th generation mobile communication system, LTE, does not support an additional core network to provide voice service, and it is merged into a packet network based on all IP. Although Voice service over LTE can be supported by VoIP, it will be provided by the existing 3G networks because of the discontinuity of LTE coverage. However, it is inevitable to adopt VoIP over LTE to provide high quality voice service. In this paper, we investigate the capacity of VoIP over LTE. Our results indicate that spectral efficiency can be significantly improved as channel bandwidth increases in terms of VoLTE capacity. In addition, we can achieve higher VoLTE capacity without decreasing control channel capacity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12B
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• pp.1677-1685
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• 2001

In this paper, we propose a call admission control scheme that can be performed within guaranteeing of required QoS(Quality of Services) in DS-CDMA(Direct Sequence-Code Division Multiple Access) reverse link. It has been performed rely on a physical channel numberonly and based on quality of received signal from MODEM(modulator/demodulator) part in established study. In other methods, the standard for services would have been set from statistical analysis of users\` location and using received power level in BTS(Base Transceiver Station). These ways bring about not only system loads but time delay or great differences from real environment. To solve these problems, we propose a call drop rate estimation algorithm for the purpose of call admission control based on measured value at LNA(Low Noise Amplifier) ports of BTS(Base Transceiver Station) in real time. This method proposed in this paper estimates a quality of offered service in real time, reduce system loads and shorten time delay which is needed to determine the standard for call admission control. But it requires a additional 17W complexity which can measure received signal power in BTS and estimate call drop rate.

• Convergence Security Journal
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• v.13 no.4
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• pp.19-25
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• 2013

The increasing demand for fast, flexible and guaranteed Quality of Service (QoS) in core networks has caused to deploy MultiProtocol Label Switching (MPLS) and Generalized MPLS (GMPLS) control plane. In GMPLS control plane, path computation and cooperation processes are one of the crucial element to maintain an acceptable level of service. The Internet Engineering Task Force (IETF) has proposed the Path Computation Element (PCE) architecture. The PCE is a dedicated network element devoted to path computation process and communications between Path Computation Clients (PCC) and PCEs is realized through the PCE Protocol (PCEP). This paper examines the PCE-based path computation architecture to include the design and implementation of PCEP. The functional modules including Finite State Machine (FSM) and related key design issues of each state are presented. In particular we also discuss internal/external protocol interfaces that efficiently control the communication channels.

• Journal of the Korea Society of Computer and Information
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• v.19 no.3
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• pp.63-71
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• 2014

We analyze the execution time of Reed-Solomon coding, which is the MAC-layer forward error correction scheme used in CDMA2000 1xEV-DO broadcast services, under different air channel conditions. The results show that the time constraints of MPEG-4 cannot be guaranteed by Reed-Solomon decoding when the packet loss rate (PLR) is high, due to its long computation time on current hardware. To alleviate this problem, we propose three error control schemes. Our static scheme bypasses Reed-Solomon decoding at the mobile node to satisfy the MPEG-4 time constraint when the PLR exceeds a given boundary. Second, dynamic scheme corrects errors in a best-effort manner within the time constraint, instead of giving up altogether when the PLR is high; this achieves a further quality improvement. The third, video-aware dynamic scheme fixes errors in a similar way to the dynamic scheme, but in a priority-driven manner which makes the video appear smoother. Extensive simulation results show the effectiveness of our schemes compared to the original FEC scheme.