• ETRI Journal
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• 제38권6호
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• pp.1064-1073
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• 2016

An adaptive speech streaming method to improve the perceived speech quality of a software-based multipoint control unit (SW-based MCU) over IP networks is proposed. First, the proposed method predicts whether the speech packet to be transmitted is lost. To this end, the proposed method learns the pattern of packet losses in the IP network, and then predicts the loss of the packet to be transmitted over that IP network. The proposed method classifies the speech signal into different classes of silence, unvoiced, speech onset, or voiced frame. Based on the results of packet loss prediction and speech classification, the proposed method determines the proper amount and bitrate of redundant speech data (RSD) that are sent with primary speech data (PSD) in order to assist the speech decoder to restore the speech signals of lost packets. Specifically, when a packet is predicted to be lost, the amount and bitrate of the RSD must be increased through a reduction in the bitrate of the PSD. The effectiveness of the proposed method for learning the packet loss pattern and assigning a different speech coding rate is then demonstrated using a support vector machine and adaptive multirate-narrowband, respectively. The results show that as compared with conventional methods that restore lost speech signals, the proposed method remarkably improves the perceived speech quality of an SW-based MCU under various packet loss conditions in an IP network.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제13권3호
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• pp.1298-1310
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• 2019

Network coding (NC) is a promising technology that can improve available bandwidth and packet throughput in wireless sensor networks (WSNs). Sliding window is an improved technology of NC, which is a supplement of TCP/IP technology and can improve data throughput and network lifetime on WSNs. This paper proposes a network coding-based maximum lifetime algorithm for sliding window in WSNs (NC-MLSW) which improves the throughput and network lifetime in WSN. The packets on the source node are sent on the WSNs. The intermediate node encodes the received original packet and forwards the newly encoded packet to the next node. Finally, the destination node decodes the received encoded data packet and recovers the original packet. The performance of the NC-MLSW algorithm is studied using NS2 simulation software and the network packet throughput, network lifetime and data packet loss rate were evaluated. The simulations experiment results show that the NC-MLSW algorithm can obviously improve the network packet throughput and network lifetime.

• 한국정보처리학회논문지
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• 제6권11S호
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• pp.3190-3198
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• 1999

Because of the rapid growth of the mobile communication, the need for the mobile internet access has grown up as well. since the current mobile communication network, however, is optimized for a voice communication system, which exclusively occupies a channel for a given time, it is not suitable for variable rate packet data. In order to support the mobile internet access, it is essential do design a reasonable packet switching network which supports the mobility. Since mobile packet network has longer latency, high speed switching and QoS are required to meet the user's requirements. In this paper, we suggest an resonable way to construct a network and its operation procedures utilizing GPRS(General Packet Radio Service) network and MPLS(Multi Protocol Label Switching) to provide a high speed switching and QoS mobile internet access. GPRS is used as a network which supports the mobility and MPLS guarantees the QoS and high speed IP protocol transmission based on the ATM switching technology.

• 한국위성정보통신학회논문지
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• 제5권2호
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• pp.97-103
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• 2010

본 논문에서는 S-대역 위성/지상 겸용 다중빔 위성통신시스템에 적용을 위한 효율적인 기저대역 패킷 스위치 설계 방안을 제시하기 위하여 설계 요구사항과 고려사항들을 분석하고 기존의 기저대역 패킷 스위칭 기법들의 요소기술과 특성을 비교 분석하였다. 위성 기저대역 패킷 스위치의 설계 요구사항 정립을 위하여 기저대역 패킷 스위치의 요구기능, 시스템 규격, 통신서비스 종류와 성능요구사항을 분석하였고, 기존의 레이어-2 및 레이어-3 패킷 스위치 특성, 스위칭 프로토콜 특성, 패킷 스위치 구조 특성들에 대한 비교 분석 결과를 기반으로 MPEG 기반의 레이어-2 위성 기저대역 패킷 스위치의 하드웨어 설계 방안을 제시하였다.

• Journal of Communications and Networks
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• 제5권1호
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• pp.25-32
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• 2003

4G mobile communication system requires the throughput of 10-100Mbps. Adaptive modulated OFDM system is promising technique for increasing the throughput. In the pilot symbol assisted high-speed packet transmission system, the data symbol duration is generally considered to be small compared to the coherence time. However, OFDM symbol duration is longer than the symbol duration of a single carrier system, so that the packet duration of the pilot symbol assisted high speed packet transmission system is long. In this case, the change of channel conditions is too fast to be accurately estimated by channel estimator at the receiver in high Doppler frequency, so that many errors occur during demodulation, especially with the data symbols at the end of each packet. In this paper, we consider the BER at various instantaneous $E_b/N_o$ that includes the demodulation errors in high Doppler frequency. When the coherence time is ten times longer than the duration of a single packet, the channel can be closely approximated as an AWGN channel. Otherwise, the approximation breaks down and the above-mentioned errors that occur during demodulation must be taken into consideration. In this paper, we propose the pilot symbol assisted high speed packet transmission system based on adaptive OFDM using a novel lookup table to consider the demodulated errors and evaluate the throughput performance.

• 한국인터넷방송통신학회논문지
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• 제12권3호
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• pp.1-7
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• 2012

무선 통신 시스템의 새로운 패러다임인 중의 하나인 무선 인지 시스템에서 다양한 종류의 멀티미디어 트래픽 지원이 예상된다. 2차 사용자들이 요구하는 서비스 품질을 만족하기 위하여, 패킷 우선권 기반의 정적 자원할당 기법이 고려될 수 있다. 하지만, 이 기법은 높은 우선권을 갖는 응용 서비스의 서비스 품질을 쉽게 만족시킬 수 있으나, 낮은 우선권을 갖는 응용 서비스의 서비스 품질은 저하될 수 있다. 이에 본 논문에서는 퍼지 이론 기반의 동적 패킷 스케줄링 알고리즘을 제안한다. 제안된 기법에서는 동적 패킷 스케줄러가 각 패킷의 우선권과 지연 마감 시간(delay deadline)을 입력으로 갖는 퍼지 규칙에 따라, 각 패킷의 우선권을 동적으로 변경하여 패킷 손실율을 최소화하는 관점에서 기 사용자 채널을 통해 다음 가용한 time slot에 어떤 2차 사용자가 데이터를 전송할 지를 결정한다. 시뮬레이션을 통해 제안된 알고리즘이 우선권 기반의 정적 자원할당기법 보다 패킷 손실율을 더 향상 시킬 수 있음을 보였다.

• 안보군사학연구
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• 통권1호
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• pp.345-360
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• 2003

To build an information oriented armed forces, the Korean military telecommunication networks adopt TCP/IP standard communication infrastructures based on ATM packet switched networks. Utilizing this network infrastructure, the Korean armed forces also applies to the areas of battleship management for efficient operation command controls and resource management for efficient resource allocations. In this military communication networks, it is essential to determine the least cost network topology under equal performance and reliability constraints. Basically, this type of communication network design problem is known in the literature as an NP Hard problem. As the number of network node increases, it is very hard to obtain an optimal solution in polynomial time. Therefore, it is reasonable to use a heuristic algorithm which provides a good solution with minimal computational efforts. In this study, we developed a simulated annealing based heuristic algorithm which can be utilized for the design of military communication networks. The developed algorithm provides a good packet switched network topology which satisfies a given set of performance and reliability constraints with reasonable computation times.

• Journal of Communications and Networks
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• 제17권3호
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• pp.286-295
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• 2015

Avionic databuses fulfill a critical function in the connection and communication of aircraft components and functions such as flight-control, navigation, and monitoring. Ethernet-based avionic databuses have become the mainstream for large aircraft owning to their advantages of full-duplex communication with high bandwidth, low latency, low packet-loss, and low cost. As a new generation aviation network communication standard, avionics full-duplex switched ethernet (AFDX) adopted concepts from the telecom standard, asynchronous transfer mode (ATM). In this technology, the switches are the key devices influencing the overall performance. This paper reviews the avionic databus with emphasis on the switch architecture classifications. Based on a comparison, analysis, and discussion of the different switch architectures, we propose a new avionic switch design based on a time-division switch fabric for high flexibility and scalability. This also merges the design concept of space-partition switch fabric to achieve reliability and predictability. The new switch architecture, called space partitioned shared memory switch (SPSMS), isolates the memory space for each output port. This can reduce the competition for resources and avoid conflicts, decrease the packet forwarding latency through the switch, and reduce the packet loss rate. A simulation of the architecture with optimized network engineering tools (OPNET) confirms the efficiency and significant performance improvement over a classic shared memory switch, in terms of overall packet latency, queuing delay, and queue size.

• 한국통신학회논문지
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• 제18권10호
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• pp.1508-1517
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• 1993

본 논문에서는 분산 무선망에서 CSMA프로토콜을 사용한 무선통신 시스템을 모델링하고, 컴퓨터로 시뮬레이션을 수행하여 패킷의 도착율(arrival rate) 함수에 의한 평균 END-to-END 지연특성을 살펴보았고, 또 모델링 을 수학적으로도 해석하였다. 이 무선 통신시스템은 단방향 통신방식으로 비동기식 1-persistent CSMA프로토콜을 사용하였으며, 또 모든 무선 통신시스템들이 근거러 영역에 적절하게 위치하고 있다고 가정하였고, 트래픽이 공용채널에 포아송 분포를 이루고 있다고 가정하였다. 이 모델의 해석에서는 무선 통신시스템이 breakdown 특성을 갖는 M/D/1 모델을 기본으로 하였다. 결과적으로 본 논문의 모델에 기초한 무선망은 패킷도착율이 2(packet/sec) 미만인 경우 평균지연시간이 패킷전송시간의 2배정도로 잘 동작됨을 확인할 수 있다.

• Journal of Communications and Networks
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• 제11권1호
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• pp.36-41
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• 2009

The IEEE 802.16 wireless metropolitan area network (WMAN) standard is designed to correct expensive communication costs in CDMA-based mobile communication systems and limited coverage problems in wireless LAN systems. Thus, the IEEE 802.16e standard can provide mobile high-speed packet access between mobile stations and the Internet service provider through the base station with cheap communication fees. To efficiently accommodate voice over IP (VoIP) services in IEEE 802.16 systems, an uplink quality of service packet-scheduling scheme is proposed, and its performance is evaluated with an NS-2 network simulator in this paper. Numerical results show that this proposed scheme can increase the system capacity by 100% more than in the unsolicited rand service (UGS) scheme and 30% more than the extended real-time polling service (ertPS) scheme, respectively.