• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.7A
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• pp.964-972
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• 1999

In this paper, we proposed a simulation model to which apply the AAL 2 (AAL type 2) between BSC and MSC in the cellular mobile communication systems. We suggested the frame structure of processing the packets of short length and the scheme which multiplex to one or more ATM cell. Also, we analyzed the performance of the APR, transmission delay, and channel transmission efficiency used in the packetized voice traffic and the low bit-rate data traffic such as fax. From the simulation results, the maximum number of users are 47 users without using AAL 2 multiplexing, but the maximum number of users are 70 (Non-Overlapping scheme) users, 110 (Overlapping scheme) users, respectively. Thus, we knew that the Overlapping scheme is more efficient than the Non-Overlapping scheme. Finally, we showed that the optimum transmission buffer size is 4 ATM cells in the cellular communication systems with the bandwidth of 2 Mbps.

• ETRI Journal
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• v.40 no.3
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• pp.318-329
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• 2018

Femtocell (FC) technology envisaged as a cost-effective approach to attain better indoor coverage of mobile voice and data service. Deployment of FCs over macrocell forms a heterogeneous network. In urban areas, the key factor limits the successful deployment of FCs is inter-cell interference (ICI), which severely affects the performance of victim users. Autonomous FC transmission power setting is one straightforward way for coordinating ICI in the downlink. Application of intelligent control using soft computing techniques has not yet explored well for wireless networks. In this work, autonomous FC transmission power setting strategy using Adaptive Neuro Fuzzy Inference System is proposed. The main advantage of the proposed method is zero signaling overhead, reduced computational complexity and bare minimum delay in performing power setting of FC base station because only the periodic channel measurement reports fed back by the user equipment are needed. System level simulation results validate the effectiveness of the proposed method by providing much better throughput, even under high interference activation scenario and cell edge users can be prevented from going outage.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.12 no.5
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• pp.2305-2310
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• 2011

Recently, communication industry based on data and voice and broadcasting industry centering around images have been rapidly blended. Thereupon, this article aims to suggest a multi-approach method which minimizes the use of network bandwidth allowing multimedia streaming transmission to secure IP-based continuity and let users get multimedia services of one channel or several simultaneously. Also, this study intends to design a buffering strategy that can absorb network delay and an object model to assign and maintain stable channel bandwidth.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.2
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• pp.279-287
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• 2001

In this paper, we discuss the scale of subjective speech quality measurement over VoIP(Voice over IP) network which is a component of broadband networks. Objective parameters of multimedia services like PSNR or jitter can easily measured and defined, but these factors are not easily meet the user's perceptual recognition. We suggest the speech quality measurement scale through the subjective measurement for end-to-end speech quality composed of sender-side quality, transmission quality, receiver-side quality, which provide the degree of correctness of representation of speaker, the degree of impairment caused by various factors, the degree of recognition of processed speech, respectively. Also, we examined the proposed method and verify it's availability.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.1A
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• pp.96-103
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• 2000

The voice traffic is one of the most important real-time objects in WATM(Wireless Asynchronous Transfer Mode) networks. In this paper, we propose a new MAC(Medium Access'Control) protocol for the efficienttransmission of voice traffic over WATM networks in the multimedia environment and compare the performanceto existing similar protocols. The new protocol separates the reservation slot period for voice and that for data toguarantee some level of QoS(Quality of Service) in voice traffic. This is denoted by a slot assignment functiondepending on the frame size. According to the characteristics of voice traffic which is repeatedly in silent states,the protocol allocates voice reservation request slots dynamically with respect to the number of silent(off state)voice sources and also sends the first block of talkspurt restarted after silent period with a reservation requestslot to reduce the access delay.The simulation results show that the proposed protocol has better performance than Slotted Aloha in bandwidthefficiency, and can serve a certain level of QoS by the given slot assignment function even when the number ofvoice terminals varies dynamically. This means we can observe that the new MAC protocol is much better thanthe NC-PRMA(None Collision-Packet Reservation Multiple Access) protocol.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.37 no.5
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• pp.36-47
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• 2000

The CDMA digital cellular systems provide better voice Quality than analog systems, however there exists inherent delays due to speech coding and transmission processing, which brings echoes returned by the BSC and PSTN interface. In this paper, we show the performance improvement of a proposed echo canceller by real time implementation, where Block Update NLMS algorithm is applied into the TMS320C54X DSP. By applying the proposed method into the practical mobile phone, we verify that various types of echoes (LE, ESE, AE) may be removed more precisely. We also cope with echo path change resulting from change of delay length after taking VAD to find echo path delay.

• Proceedings of the Korean Operations and Management Science Society Conference
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• 2006.05a
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• pp.1656-1660
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• 2006

Distributed coordination function(DCF) is the primary random access mechanism of IEEE 802.11, which is the basic protocol of wireless LAN based on the CSMA/CA protocol. It enables fast installation with minimal management and maintenance costs and is a very robust protocol for the best effort service in wireless medium. The current DCF, however, is known to be unsuitable for real-time applications such as voice message transmission. In this paper, we focus on the performance issues of IEEE 802.11 which accommodate the prioritized messages. Existing results use the initial window size and backoff window-increasing factor as tools to handle the priority of the messages. Instead, we introduce a novel scheme which chooses the backoff timer with arbitrary probabilities. By this, one can greatly reduce the backoff delay of the lower priority messages without degrading the performance of higher priority.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.05a
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• pp.366-372
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• 2002

Tele-medicine and emergency medical system are necessary for moving from an accidental point or far distance to a hospital and emergency treatment or home treatment before a hospital. Emergency treatment is extremely important in the case of death before arriving a hospital and deformed or disabled by medical treatment delay. A necessary element for this medical system is the emergency communication system. This system is on preparing for an ability of furnishing patient status to a corresponding health service by monitoring the patient at an ambulance of the accident place. This is the transportation of basic biological information of a patient to a medical center by wireless communication system and the corresponding hospital or medical center examine the patient by monitoring, then they can send emergency medical order to the patient for emergency treatment. The TRS is most efficient way of emergency medical communication system, which is currently used with popularity. In this paper studied simultaneously a way of detecting and transporting bio-logical signals, and monitoring of transporting data with communication of voice in the accident place or ambulance.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.7-10
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• 2000

본 논문은 이동통신상에서 송신측의 송신 채널 에러정보를 수신측에서 송신측으로 전달하는 음성패킷을 이용하여 송신측에 알려주어 압축과정에 이용하게 할 수 있는 방법을 제안한다. ACELP(Algebraic CELP)구조 방식을 가지고 있는 음성압축기들 중 G.729을 사용하며 음성 패킷정보 중 Pitch Delay와 Fixed Codebook를 이용하여 전송음성 패킷안에 상대방의 송신 채널정보를 싣는다. 수신측에서 받은 패킷이 Erasure로 판단되었을 때 패킷정보들 중 Fixed Codebook Index를 만들게 되는 4개의 Optimal 펄스 중 2개의 펄스만 사용하며 나머지 2개의 펄스는 약속된 임의의 위치에 위치시킨 후 송신측에 전송시킨다. 상대방에서 약속에 맞는 위치의 펄스를 보내왔을 때를 체크함으로써 자신의 송신채널 상태를 알 수 있게 된다 송신채널에$5\%$ Erasure 채널 에러가 발생했을 때 채널정보를 가진 패킷의 음질은 약 0.1dB 정도 떨어지게 된다. 하지만 음성압축 전송 시 송신채널의 정보를 이용하여 무선채널에러에 강하게 할 수 있다.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.3
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• pp.103-110
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• 2010

This paper addressed the implementation of the systematic performance analysis of Quality of Service (QoS) by using OPNET simulator in the interworking architecture of IEEE 802.16e (mobile WiMAX) and IEEE 802.11e (WLAN) wireless network. Four simulation cases were provided in OPNET simulator and a voice traffic was simulated with various performance metrics, such as Mean Opinion Score (MOS), end-to-end delay and packet transmission ratio. Based on the simulation results, the MOS value presented better in WiMAX to WiMAX case compared to others in both static and mobility case. Meanwhile, end-to-end delay was not greatly affected by mobility in four cases. However, mobility was affected much in MOS value and packet transmission ratio in WLAN to WLAN case than in others.