• Journal of the Institute of Electronics Engineers of Korea CI
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• v.41 no.3
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• pp.47-54
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• 2004

By using wireless terminal, the number of users who wish to use the multimedia service like the Internet as well as Short Message Services and voice service has increased dramatically over the last years. We propose the method that improves Mobile IPv4 (MIPv4) Regional Registration in wireless mobile networks to decrease traffic's transmission delay and message generation compared with an existing method We design the scheme in MIPv4 environments that a packet do not pass through the home agent transmitted from correspondent node to mobile node, if a mobile node moves to other mobility agent. Simulation results show that the proposed method significantly reduces the expenses for registration and delivering packet.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.1
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• pp.259-266
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• 2013

Since the wireless communications have a limited transmission, the devices just around a master node can exchange data. Though Bluetooth and Zigbee support ad hoc, they are not appropriate for real-time voice communications. In this paper, we present a TDMA-based relay protocol for several users to communicate simultaneously. The proposed protocol can relay data or voice to other nodes in real-time by the multi-hop transmission method using TDMA. And the proposed protocol improves the network performance by allocating different frequencies to the slaves depending on the routing path scheduled by the routing table. NS-2 simulation shows that the performance of the proposed protocol is good in terms of the transmission delay and pecket loss probability in the real-time voice transmission.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.12B
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• pp.1988-1995
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• 2000

ATM 망에서 제공되고 있는 음성, 영상, 데이터와 같은 다양한 서비스는 사용자의 만족도를 수용할 수 있어야 한다는 것을 전제조건으로 한다. 이러한 기본적인 요구사항을 충족시키기 위해서는 노드대 노드간의 자원관리와 오류제어 및 다양한 트래픽의 특성을 고려한 전송 순서의 결정에 해당하는 스케줄링 방법이 요구되어 진다. 본 논문에서는 이러한 기술 요소 중에서 트래픽 설정 단계에서 제공되는 트래픽 특성 및 QoS(Quality of Service) 정보를 바탕으로 교환 노드에서 발생되는 전달 지연 시간에 적응적인 스케줄링 방식을 제시하였다. 이 방식은 멀티미디어 서비스와 같이 혼합된 트래픽 특성을 갖는 구조에서 트래픽 구성비율에 따라 적용되는 지연 여유치를 매우 융통성 있고 효과적으로 조절할 수 있다. 성능분석을 위하여 기존의 스케줄링 방식인 WFQ (Weighted Fair Queueing) 방식과 제안한 스케줄링 방식의 수학적인 분석을 수행하였으며 이 두 방식의 결과식을 비교하여 교환노드에서의 평균 지연과 셀 처리에 관한 성능을 분석하였다. 그리고 수학적 분석에 대한 검증으로는 Simulation tool ARENA 3.0을 이용하여 제안한 알고리즘의 Worst case와 기존의 알고리즘의 성능을 비교하였다.

• The Journal of the Acoustical Society of Korea
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• v.33 no.2
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• pp.133-138
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• 2014

Packet arrival-delay variation, so-called 'jitter' is one of the main factors that degrade the quality of voice in mobile devices at the Voice over Internet Protocol (VoIP). To resolve this issue, a playout scheduling based on adaptive jitter estimation for enhancing VoIP speech quality is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. The experimental results have shown that the proposed algorithm delivers high voice quality in unstable network environment.

• Journal of the Institute of Convergence Signal Processing
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• v.10 no.1
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• pp.1-6
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• 2009

Propagation of electromagnetic wave in the water or underground is very difficult because of the conductivity of the propagation materials. In this case, we usually use acoustic signal as ultrasonic but, it is not easy to transfer long distance with coherent method because of time varying multipath, doppler effect, and attenuations. So, we use noncoherent method as FSK to communicate between long distances. But, as the propagation speed of acoustic sound is very slow, the BW of the channel is narrow. It is very hard to guaranty the enough speed of communication like digital image data. In this paper, we proposed a new data communication protocol which can transmit multi-bit digital data with every single ping, and improve the data communication speed in the water.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.12A
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• pp.1852-1861
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• 1999

In this paper, CDMA Reservation ALOHA scheme which can reduce multiple access interference and packet collision is proposed to support multi-traffic such as voice and random data with and without priority. In this scheme, the time slot is divided into two stage, access and transmission stage. Only packets with spreading codes assigned from base station in access stage can transmit their packets in transmission stage, so MAI can be reduced. To reduce packer collision in access stage, the code reservation and access permission probability are used. Code reservation is allowed for voice traffic and continuous traffic with priority using piggyback and access permission probability based on the estimation of the number of contending users in the steady-state is adaptively applied to each traffic terminal. Also, we analyzed and simulated the numerical performances required for each traffic using Markov chain modeling in a single cell environment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.11A
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• pp.1106-1112
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• 2007

As a demand for high speed uplink packet services increases, the WCDMA enhanced uplink, also known as high speed uplink packet access (HSUPA), has been specified in release 6 by 3GPP. This HSUPA will provide various types of multimedia services, such as real-time video streaming, gaming, VoIP, and FTP. Generally, the performance of HSUPA is dominated by scheduling policy. Therefore, it is required to design a scheduling algorithm considering the traffic characteristics to provide QoS guaranteed services in various traffic environments. In this paper, we propose a scheduling algorithm considering the traffic characteristics to guarantee QoS in a mixed traffic environment. Finally, the performance of the proposed scheduling algorithm is evaluated in terms of average packet delay, packet delay jitter, and system throughput using a system level simulator.

• The Journal of the Acoustical Society of Korea
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• v.18 no.7
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• pp.33-38
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• 1999

In this paper, we combine a generalized analysis-by-synthesis (AbS) structure and an algebraic excitation scheme to propose a new 4kb/s speech codec. This codec partly uses the structure of G.729. We design a line spectrum pair (LSP) quantizer, an adaptive codebook, and an excitation codebook to fit the 4 kb/s bit rate. The codec has a 25㎳ algorithmic delay, which corresponds to a 20㎳ frame size and a 5㎳ lookahead. At the bit rates below 4kb/s, most CELP speech codecs using the AbS principle have a drawback that results a rapid degradation of speech quality. To overcome this drawback we use the generalized AbS structure which is efficient for the low bit rate speech codec. LP coefficients are converted to LSP and quantized using a predictive 2-stage VQ. A low complexity algebraic codebook which uses shifting method is used for the fixed codebook excitation, and gains of the adaptive codebook and the fixed codebook are quantized using the VQ. To evaluate the performance of the proposed codec A-B preference tests are done with the fixed rate 8kb/s QCELP. As the result of the test, the performance of the codec is similar to that of the fixed rate 8kb/s QCELP.

• Proceedings of the Korea Society for Industrial Systems Conference
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• 2000.11a
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• pp.463-470
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• 2000

In this paper, we describe a wireless MAC protocol named APRMA(Abitrary Period Reservation Multiple Access), which is capable of supporting the ABR type data service and maximizing channel utilization. In original PRMA protocol, data terminals with random data packets cannot reserve slot. That is, slot reservation is applicable to the. time constraint voice packet exclusively. But the reservation scheme have to be performed for loss sensitive data packet, so data packets can get their quality of service. The aspects of service, if fixed bandwidth is allocated to data terminals, time constraint voice packets may have a low efficiency So in this study, the terminal which wants to request for ABR type service, acquires a minimum bandwidth from system for the first time. If the system have extra available bandwidth, ABR terminals would acquire additional bandwidth slot by slot. As a result, APRMA protocol can support the data service with loss sensitivity and maintain their channel utilization highly. Also high Priority services like voice can be satisfied with their QoS by APRMA.

• Proceedings of the KIEE Conference
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• 2008.10b
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• pp.503-504
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• 2008

최근 정보통신 기술들이 빠르게 발전하고 있다. 다양한 통신 기술들 중에서도 업무의 효율을 높이고자 회사 및 가정, 학교 등에서 자주 사용되고 있는 영상채팅시스템을 구현해보고자 한다. 쿼타임 코덱 중 가장 보편적인 코덱으로 인코딩이 쉽고 저사양의 CPU만으로도 실시간 스트리밍이 가능한 H.263 코덱을 사용하여 영상채팅시스템을 Visual C++로 구현을 하였다. 전송로의 지연을 줄이기 위하여 영상, 음성, 텍스트 등을 압축하고 복원하는 데 걸리는 시간을 최소화기 위하여 데이터의 전송대역폭을 적절히 조절하는 알고리듬을 제안하여 전송지연을 최소화하였다. 또한 P2P 방식을 사용하여 다양한 영상 환경에 대하여 영상 및 텍스트 데이터의 안정성과 화질이 우수함을 보였으며, 실시간 가스안전관리 상담에 이용하여 업무의 효율을 높이고자 한다.