• Journal of Internet Computing and Services
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• v.5 no.6
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• pp.31-43
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• 2004

Voice over Internet Protocol (VoIP) is currently a popular research topic as a real time voice packet transmission method. But current Internet environment do not guarantee the quality of voice when we take a side view of delay, jitter and loss. Up to now, many voice based evaluation algorithms have been used to measure speech quality of VoIP systems. However, these algorithms have the defects that their results are different according to voice samples and some algorithms can not take network environment for speech transmission path. The E-model can be used to solve the problems of these algorithms. In this paper. we introduce VoIP planning guidelines through the various analysis of E-model which can model impairments of network quality as well as VoIP equipment quality systematically, We, also, show the evaluation method and results of speech transmission quality.

• Proceedings of the Korean Operations and Management Science Society Conference
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• 2007.11a
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• pp.325-329
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• 2007

The mobile and multimedia service on wireless network have been leaded from the improvement of telecommunication techniques. A typical multimedia service, a video transmission usually requires larger bandwidth than voice transmission. Many channel allocation algorithms for Orthogonal Frequency Division Multiple Access (OFBMA) to use resources more efficiently, Previous channel allocation algorithms have developed with an assumption that the data traffic is constant bit rate (CBR). However, existing algorithms are not suitable to video traffic because it usually generates a variable bit rate (VBR) traffic. In this paper, we proposed a new channel allocation algorithm called a queue-based channel allocation. it is more suitable to transmit the video traffics. Also, a problem are notified in case of realtime generated video traffic and a corresponding heuristic solution was proposed.

• Proceedings of the IEEK Conference
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• 2000.06a
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• pp.225-228
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• 2000

In this paper, a design of the reliable video-conference system using CTI(Computer Telephony Integration) technology is proposed. When video-conference is run on the current existing Internet, the transmission delay problem for voice data traffic can be frequently occurred. In order to transmit the real-time voice data through the Internet efficiently, some complicated algorithms such as CODEC(Code/Decode) should be applied. It can cause further excessive processing delay which can affect the overall performance. The voice traffic is usually transmitted through the reliable PSTN(Public Switched Telephone Network) in the CTI system. In this paper a new architecture, in which PSTN for voice traffic and Internet for video traffic are used at the same time instead of using Internet by itself, is proposed to relieve the problems on a video conference.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.42 no.2 s.302
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• pp.11-20
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• 2005

In previous video conference system, when the number of participants in video conference increases by n, the bandwidth and memory of n2 is required. And also, it brings about increase in traffic and problem of a say during a conference in aspect of transmission of voice data. In this paper, we propose an algorithm of remote video conference using silence detection algerian to resolve the questions such as buffering method of video data in server and heavy traffic detection algorithm to the increase in participants. Video data buffering algorithm is not a method of broadcasting to other client in the server, but this algorithm uses two other methods; the buffering method of receiving compressed video data from clients and the indexing method for acquiring the video data of other participants in clients according to clients' bandwidth and network transmission speed. We apply a voice transmission algerian and a channel management algorithm to the remote video conference system. The method used in the voice transmission algorithm is a silence detection algorithm which does not send silent participants' voice data to the server. The channel management algorithm is a method allocating a say to the participants who have priority. In consideration of average 20 frames and 30ms regardless of a number of participants, we can safely conclude that the transmission of video and voice data is stable.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.9
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• pp.20-27
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• 2002

The demand for multimedia transmission in wireless networking is rapidly increasing. Performance evaluation in the CDMA system has been carried out on the voice-oriented system without re-transmission or buffering. We propose a multimedia access protocol with buffering and ARQ which can meet a variety of QoS requirements. We study the effect of buffering in MC-CDMA(Multi-Code CDMA) and design a dynamic rate control algorithms, whose simulation shows the efficiency of MC-CDMA.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.2
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• pp.307-312
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• 2000

Asynchronous transfer mode(ATM) is flexible to support the various multimedia communication services such as data, voice, and image by applying asynchronous time-sharing and statistical multiplexing techniques to the existing data communication. ATM service is categorized to CBR, VBR, UBR, and ABR according to characteristics of the traffic and a required service qualities. Among them, ABR service guarantees a minimal bandwidth and can transmit cells at a maximum transmission rate within the available bandwidth. To minimize the cell losses in transmission and switching, a feedback information in ATM network is used to control the traffic. In this paper, predictive control algorithms are proposed for the feedback information. When the feedback information takes a long propagation delay to the backward nodes, ATM switch can experience a congestion situation from the queue length increases, and a high queue length fluctuations in time. The control algorithms proposed in this paper provides predictive control model using slop changes of the queue length function and previous data of the queue lengths. Simulation shows the effectiveness result of the proposed control algorithms.

• Proceedings of the Korean Operations and Management Science Society Conference
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• 1997.10a
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• pp.245-248
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• 1997

As the increasing the demand on information service, high-capacity and high-speed telecommunication networks are required. For the networks, very intelligent telecommunication equipments such as DCS(Digital Cross-connect System) will be employed for the fast service on the various types of information including voice, data and image. This paper considers the transmission networks composed of DCSs and optical fibers as nodes and links of the networks, respectively. For the networks, some types of restoration algorithms are compared their characteristics for their potential applications. And a distributed control algorithm is described as an empirical example which is implemented on the BDCS(Broadband Digital Cross-connect System), where the BDCS is a type of DCS developed in Korea. Finally, some remarks on the associated further researches are added.

• Journal of Digital Contents Society
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• v.3 no.1
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• pp.47-57
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• 2002

Recently, wireless networks have been pursuing multimedia data service as voice, data, image, video and various form of data according to development of information communication technology. It guarantees cell delivery delay of real time data in efficient real time multimedia data transfer. Also, it minimizes cell loss rate of non-real time multimedia data. In the wireless ATM, there are based on Asynchronous Transfer Mode(ATM). It implies that there are various service with difficult transmission rates and qualities in the wireless communication network. As a result, it is important to find out the ways to guarantee the Quality of Service(QoS) for each kind of traffic in wireless network. In this thesis, we propose an improved TCRM scheduling algorithms for transmission real-time multimedia data service in wireless ATM Networks. We appear real time multimedia scheduling policy that apply each different method to uplink and downlik to wireless ATM network. It can guarantee QoS requirements for each real time data and non-real time data. It also deals the fairness problem for sharing the scarce wireless resources. We solve fault of TCRM as inefficient problem of non-real data by using arbitrary transmission speed and RB(Reservation Buffer) through VC(Virtual Control) and BS(Base Station).

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.6
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• pp.36-45
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• 1998

This paper discusses multiple access control aspects for a wireless extension of high-speed asynchronous transfer mode(ATM) networks. One challenging requirement is that mobile systems convey of diverse types of information including voice, computer data, facsimile, and video data that have different quality of service constraints. Basically, a centralized control scheme for a coordinated multiple access is required in order to efficiently accommodate multimedia traffic on wireless links. The proposed scheme dynamically scheduling the mobile terminal, which has data to transfer according to the service, types at base station. In this scheme, the scheduling algorithm is performed on the basis of traffic descriptors and QoS parameters, which required in the connection set up phase. As a consequence, the transmission scheduling is very flexible and can account for the different traffic rate and delay constraints that emerge from voice and data integration. The simulation result shows that the proposed scheme has better performance over other scheduling algorithms.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.4
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• pp.782-792
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• 1997

Asynchronous transfer mode (ATM) provides the means to support with variety of multimedia serivices in broadband network. For the support of multimedia services, various data traffic with priortes should be controlled errectively and congestion function is required in the network to carry out the control operation. To accomplish this in a flexible and effective manner, priorty classes for the different services are commonly used. In this paper, we suggested a buffer managment algorithm in ATM network to improve overall network performance with separated buffer which consists of existing buffer with threshold andauxiliary buffer whose input consists of superposition of voice and data traffic. To evaluate the proposed buffer managment algorithms, simulations are excuted with two proiorities and the results are provied that network throughput is improved better than the existing parital buffer method.