• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.48-55
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• 2010

It has a falling off in speech quality, especially when consecutive packet loss occurs, even if a vocoder implemented in the packet network has its own packet loss concealment (PLC) algorithm. PLC algorithm is divided into transmitter and receiver algorithm. Algorithm in the transmitter gives superior quality by additional information. however it is impossible to provide mutual compatibility and it occurs extra delay and transmission rate. The method applied in the receiver does not require additional delay. However, it sets limits to improve the speech quality. In this paper, we propose a new method that puts extra information for PLC in a part of Extension Header Data which is not used in RTP Header. It can solve the problem and obtain enhanced speech quality. There is no extra delay occurred by the proposed algorithm because there is a jitter buffer to adjust network delay in a receiver. Extra information, 16 bits each frame for G.729 PLC, is allocated for MA filter index in LP synthesis, excitation signal, excitation signal gain and residual gain reconstruction. It is because a transmitter sends speech data each 20 ms when it transfers RTP payload. As a result, the proposed method shows superior performance about 13.5%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.10
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• pp.2292-2302
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• 1997

In this paper, we propose a wireless MAC protocol named APRMA, which is capable of supporting the ABR type data service and maximizing channel utilization. In 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 has to be performed for loss sensitive data packet, and so data packets can get their quality of service. Therefore, in wireless MAC, reservation technique has to be used for both voice and data services. But in service aspects, if a 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 tyupe service, acquires a minimum bandwidth from system for thefirst time. If the system has extra available bandwidth, ABR terminals would acquire additional bandwidth slot by slot. As a result, APRMA protocol cansupporty the data service with loss sensitivity and maintain their channel utilization high. Also high priority services like voice can be satisfied with their QoS by APRMA.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.8B
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• pp.739-746
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• 2004

An ATM/AAL2 is standardized to transmit delay sensitive application services, which has small size packet, efficiently. An AAL2 transmission scheme is used to deliver voice and data traffic on the lob interface between base station (Node-B) and Radio Network Controller (RNC) in UMTS network. To predict AAL2 performance, a detailed end-to-end UMTS network performance simulator was developed. We performed detailed simulation(cell packing density and bandwidth gain) for voice and data services in UTRAN. The results indicate that the maximum bandwidth gain in Node-B is about 17% and the bandwidth gain of AAL2 multiplexing in $I_{ub}$ for data services is less than that for voice service. Futhermore, the more offered load increase the more the bandwidth gain decreases in a concentrator.

• Proceedings of the Korea Society of Information Technology Applications Conference
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• 2004.06a
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• pp.55-56
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• 2004

차세대 이동통신망인 B3G 시스템은 음성 트래픽뿐만 아니라 데이터, 화상, 비디오와 같은 멀티미디어 트래픽을 지원하여야 하므로 더 많은 무선 자원을 요구한다. 특히 미디어 트래픽의 전송 중에 핸드오버가 발생하면 멀티미디어 트래픽의 QoS가 지연 및 손실에 의하여 영향 받기 때문에, 정지 상태에서의 경우와 대등한 QoS를 유지하기 위해서는 효율적인 자원 예약 및 할당 방안이 필요하다. 본 논문에서는 이동 멀티미디어 망에서 이동 단말기의 이동 방향 추정에 근거하여 자원을 예약하고, 멀티미디어 트래픽을 전송하는 핸드오버 기법을 제안한다.

• The KIPS Transactions:PartC
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• v.9C no.3
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• pp.399-406
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• 2002

We consider design of multiservice network link, in which connections of constant- and adaptive-rate services arrive and leave dynamically. We propose performance analysis and design methods of these dynamic multiservice networks. A multiservice network link is modeled by a Markov chain, and data rates for adaptive-rate connections are derived using QBD (Quasi-Birth-Death) processes and matrix-geometric equations. We estimate average number of adaptive-rate connections, average data rate and average connection delay. The performance of constant-rate connections is determined from the blocking probability of the connections. Based on the performance of constant-and adaptive- rate connections, we propose a design methods of a network link to satisfy performance requirements of constant- and adaptive-rate connections (data rates, delay, blocking probability). Our methods can be used for the analysis and design of network switch supporting dynamic data and voice (video) traffic simultaneously.

• The Journal of the Korea institute of electronic communication sciences
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• v.9 no.6
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• pp.719-726
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• 2014

This paper addresses quality of experience(QoE) and how to measure and evaluate QoE including its subjective aspects. Adopting the real measurements on the field, a various quality metric have been measured for VoIP(voice over IP) service provided through a wireless interface of WiBro(Wireless Broadband). By analyzing the measured values and correlation between the metrics, we attempt to find a method to evaluate QoE of the VoIP service in a objective way. As a result, it has been shown that QoE of the VoIP service through WiBro network has close relation to the packet-level end-to-end delay, and the delay has close relation to received signal strength indicator(RSSI).

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2000.11b
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• pp.29-35
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• 2000

음성 데이터의 패킷 지터(jitter)가 심할수록 오디오 플레이어가 오디오 데이터를 자연스럽게 재생하지 못하기 때문에 사용자는 원래의 음성을 거의 알아들을 수 없게 된다. 이 문제점을 해결하기 위하여 오디오 수신자는 전송 받은 오디오 데이터를 바로 재생하지 않고 재생시간을 지연시키는 방법을 사용한다. 본 연구자의 조사에 의하면 이러한 재생시간을 지연하는 대표적인 지터 컨트롤 알고리즘으로 6가지 방식이 제안되고 있다. 그 중 세 가지는 NeVot, Vat, Open H.323 프로그램 등에 구현되어 실제로 사용되고 있다 본 논문에서는 이들 6가지의 모델의 지터 컨트롤 알고리즘의 특성을 알아보고 어느 알고리즘이 효율적인지 알아보기 위해 현재 인터넷의 성능을 파악하고 이를 기초로 제안된 6가지 알고리즘 중 어느 것이 가장 효율적인가를 파악하여 오디오의 음질을 개선하기 위한 방법을 제시하고자 한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.12A
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• pp.1844-1851
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• 1999

In this paper, we propose an effective access control scheme based on service characteristics to provide multimedia services such as voice, video and data in broadband CDMA networks. In proposed algorithm, BER characteristics of each service is controlled by different Eb/No, and data traffic is controlled by access probability of data which is determined by activity factor of voice/video traffic. And we perform simulation and analysis of proposed model. As a result, the proposed access scheme shows that the outage probability is decreased by allowing delay of data traffic.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.218-223
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• 2008

In this paper, we implement several methods to improve an audio collaboration system over IP networks, and then evaluate the performance of the implemented methods. In general, speech and audio quality degrades depending on the characteristics of IP networks such as jitter and packet loss. In order to reduce this quality degradation, we propose a lower bit rate audio delivery scheme using the MPEG-2 AAC (Advanced Audio Coding) audio codec in a viewpoint that a packet loss rate could be reduced by a smaller packet size. In addition, iLBC (Internet Low-Bitrate Codec) and the G.711 packet loss concealment algorithm defined by IEFT and ITU-T, respectively, are applied to a audio collaboration system. RAT (Robust-Audio Tool)[7] is used as a baseline platform for the implementation of the proposed methods. It is shown from the implementation that the implemented MPEG-2 AAC audio codec with a bitrate of 256 kbit/s performs as similar as the uncompressed audio quality with a bitrate of 512 kbit/s, and that iLBC and the G.711 packet loss concealment algorithm can improve speech quality when a packet loss rate is 2~10%.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.11a
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• pp.425-428
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• 2003

본 논문은 유선통신망의 ATM기술을 무선통신망에 적용한 무선 멀티미디어 액세스 환경에서 패킷에러율 및 처리율등 전송성능을 개선하기 위한 통합에러제어 구조를 제안하였다. 음성, 데이터, 영상 등의 등시성 멀티미디어 트래픽을 처리해야 하는 무선 멀티미디어 네트워크에서는 트래픽별로 QoS(Quality of Service) 목표치가 다르므로 트래픽 속성에 상관없이 일괄적으로 성능개선기법들을 적용하는 것은 또 다른 성능저하 현상을 초래한다. 대표적인 멀티미디어 네트워크인 ATM 네트워크인 경우 트래픽 속성상 CBR(Constant Bit Rate) 및 랜덤트래픽은 지연에 매우 민감하며 VBR(Variable Bit Rate) 트래픽은 어느 정도의 지연은 허용되나 데이터 전송시 매우 높은 신뢰도를 필요로 한다. 이러한 배경에서 ATM을 기반으로 하는 무선 멀티미디어 네트워크 액세스 환경에서 트래픽별 QoS 속성을 만족시킬 수 있는 통합 에러제어 구조를 제시한 후 FEC, ARQ, 인터리빙, 버퍼용량등의 적용방안을 제시하였다.