• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.4
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• pp.835-843
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• 2000

This paper presents a method to design the integrated system for supporting both multimedia contents creation and its presentation in a windows system. Especially, it describes techniques for designing an integrated editor that can be used to crate and edit various media data including continuous media such as audio and video. And it also proposes a system model for systematically integrating multimedia creation tool with multimedia presentation system where the buffering and scenario-based presentation methods are included in the propose system for supporting effective multimedia presentation through a network. Finally, we describe a threads processing technique based on event monitoring to satisfy needs for presentation control, synchronization control, and user's input control during a multimedia presentation.

• MALSORI
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• no.61
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• pp.75-86
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• 2007

In this paper, we propose a scalable audio coder, which has a variable bandwidth from the narrowband speech bandwidth to the audio bandwidth and also has a bit-rate from 8 to 320 kbits/s, in order to cope with the quality of service(QoS) according to the network load. First of all, the proposed scalable coder splits bandwidth of the input audio into narrowband up to around 4 kHz and above. Next, the narrowband signals are compressed by a speech coding method compatible to an existing standard speech coder such as G.729, and the other signals whose bandwidth is above the narrowband are compressed on the basis of a psychoacoustic model. It is shown from the objective quality tests using the signal-to-noise ratio(SNR) and the perceptual evaluation of audio quality(PEAQ) that the proposed scalable audio coder provides a comparable quality to the MPEG-1 Layer III (MP3) audio coder.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.7
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• pp.10-18
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• 2013

With advancing of mobile communication services and commercializing of VoLTE (Voice of LTE), it is getting to pay attention on QoS of VoLTE. This paper proposes an integrated E-model in which some factors influenced to service quality of VoIP and VoLTE based voice communication system are considered in calculating the voice quality of Wideband Codec. The model aims to calculate R value which reflects the situations of access network, network characteristics, terminals' usage and mobility. We mainly deal with the integrated E-model's structure, related algorithms and optimal parameters for VoLTE. Some experiments show that the voice quality difference between VoIP and VoiceChecker, and VoLTE and POLQA, is below 10%. With the proposed model, we can calculate the voice quality by making use of the factors directly affected to service quality and the environment of VoLTE terminal and network. As a result, we can estimate the service quality in advance, without measuring it in real wireless environment.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.772-776
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• 2004

This paper proposes an interference avoidance approach for Constraint-Based Routing (CBR) algorithm in the Multi-Protocol Label Switching (MPLS) network. The MPLS network itself has a capability of integrating among any layer-3 protocols and any layer-2 protocols of the OSI model. It is based on the label switching technology, which is fast and flexible switching technique using pre-defined Label Switching Paths (LSPs). The MPLS network is a solution for the Traffic Engineering(TE), Quality of Service (QoS), Virtual Private Network (VPN), and Constraint-Based Routing (CBR) issues. According to the MPLS CBR, routing performance requirements are capability for on-line routing, high network throughput, high network utilization, high network scalability, fast rerouting performance, low percentage of call-setup request blocking, and low calculation complexity. There are many previously proposed algorithms such as minimum hop (MH) algorithm, widest shortest path (WSP) algorithm, and minimum interference routing algorithm (MIRA). The MIRA algorithm is currently seemed to be the best solution for the MPLS routing problem in case of selecting a path with minimum interference level. It achieves lower call-setup request blocking, lower interference level, higher network utilization and higher network throughput. However, it suffers from routing calculation complexity which makes it difficult to real task implementation. In this paper, there are three objectives for routing algorithm design, which are minimizing interference levels with other source-destination node pairs, minimizing resource usage by selecting a minimum hop path first, and reducing calculation complexity. The proposed CBR algorithm is based on power factor calculation of total amount of possible path per link and the residual bandwidth in the network. A path with high power factor should be considered as minimum interference path and should be selected for path setup. With the proposed algorithm, all of the three objectives are attained and the approach of selection of a high power factor path could minimize interference level among all source-destination node pairs. The approach of selection of a shortest path from many equal power factor paths approach could minimize the usage of network resource. Then the network has higher resource reservation for future call-setup request. Moreover, the calculation of possible path per link (or interference level indicator) is run only whenever the network topology has been changed. Hence, this approach could reduce routing calculation complexity. The simulation results show that the proposed algorithm has good performance over high network utilization, low call-setup blocking percentage and low routing computation complexity.

• The Journal of the Korea Contents Association
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• v.8 no.6
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• pp.43-53
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• 2008

Hyper text transfer protocol (HTTP) over transmission control protocol (TCP) is currently used to transfer objects in the Internet. Stream control transmission protocol (SCTP), an alternative to TCP, which allows for independent delivery among streams, and can thus reduce the mean response delay of web object. We present an analytical model to find the mean response delay for HTTP over SCTP, therefore, estimate the effectiveness of SCTP over TCP. Typical TCP delay models assume the wired environment. On the contrary, the proposed model in this paper assumes the multiple packet losses and wireless environment where fast retransmission is not possible due to small window. The estimated mean response time can be used the benchmark to meet quality of service (QoS) at end-user. We validate the accuracy of our model using experiments. It is shown that the differences between the results from model and those from experimental are very small below 6 % on average. We also find that the mean response delay for HTTP over SCTP is less than that for HTTP over TCP.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9A
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• pp.1285-1294
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• 1999

In this paper we consider the performance of TCP video traffic over ABR with Long-Range Dependent VBR traffic. As compressed coded video traffics are increasing rapidly over Internet, lots of studies are being done for transmitting those traffics efficiently using limited network resources. We consider here the transmitting video service over ABR service in ATM networks, especially satellite networks. CBR or VBR services are suggested in transmitting the video traffic in ATM Forum TM 4.0. But ABR service connection, which is considered as appropriate service for data traffic, can be established with a small amount of bandwidth, MCR (Minimum cell rate). Furthermore ABR service can control the source's transmitting rate using feedback mechanism. Using this feature ABR service can be used in some applications which can control their quality of services corresponding to network loads. Compressed video sources with MPEG-2 are used for Long-Range Dependent VBR traffic here. We model the compressed video source to resemble the MPEG-2 transport streams. These compressed video traffic streams are consisted of three different frames, I-frame, P-frame, and B-frame. So when a network are overloaded, we can control the quality of service using this traffic features. TCP Traffics over ABR need large buffers in ATM switch to satisfy their QoS with background VBR traffics, which have high deviations in bandwidth. Furthermore satellite ATM networks with large feedback delay need large buffers corresponding RTT delay. The performance comparisons among EFCI and ER switch (ERICA+) switches in the network circumstances described above were shown in this paper. We also considered the case with ON-OFF VBR traffics.

• Proceedings of the Korea Information Processing Society Conference
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• 2008.11a
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• pp.1332-1335
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• 2008

In order to various service types of real time and non-real time traffic with varying requirements are transmitted over the IEEE 802.16 standard is expected to provide quality of service(QoS) researchers have explored to provide a queue management scheme with differentiated loss guarantees for the future Internet. The sides of a packet drop rate, an each class to differential drop probability on achieving a low delay and high traffic intensity. Improved a queue management scheme to be enhanced to offer a drop probability is desired necessarily. This paper considers multiple random early detection with differential drop probability which is a slightly modified version of the Multiple-RED(Random Early Detection) model, to get the performance of the best suited, we analyzes its main control parameters (max_th, min_th, max_p) for achieving the proportional loss differentiation (PLD) model, and gives their setting guidance from the analytic approach. we propose Dynamic-multiple queue management scheme based on differential drop probability, called Dynamic-VQSDDP(Variable Queue State Differential Drop Probability)T, is proposed to overcome M-RED's shortcoming as well as supports static max_p parameter setting values for relative and each class proportional loss differentiation. M-RED is static according to the situation of the network traffic, Network environment is very dynamic situation. Therefore max_p parameter values needs to modify too to the constantly and dynamic. The verification of the guidance is shown with figuring out loss probability using a proposed algorithm under dynamic offered load and is also selection problem of optimal values of parameters for high traffic intensity and show that Dynamic-VQSDDP has the better performance in terms of packet drop rate. We also demonstrated using an ns-2 network simulation.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.3 no.6
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• pp.575-596
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• 2009

Facing limited network resources such as bandwidth and processing capability, the Internet will have congestion from time to time. In this paper, we propose a scheme to maximize the total utility offered by the network to the end user during congested times. We believe the only way to achieve our goal is to make the scheme application-aware, that is, to take advantage of the characteristics of the application. To make our scheme scalable, it is designed to be class-based. Traffic from applications with similar characteristics is classified into the same class. We adopted the RED queue management mechanism to adaptively control the traffic belonging to the same class. To achieve the optimal utility, the traffic belonging to different classes should be controlled differently. By adjusting link bandwidth assignments of different classes, the scheme can achieve the goal and adapt to the changes of dynamical incoming traffic. We use the control theoretical approach to analyze our scheme. In this paper, we focus on optimizing the control on two types of traffic flows: TCP and Simple UDP (SUDP, modeling audio or video applications based on UDP). We derive the differential equations to model the dynamics of SUDP traffic flows and drive stability conditions for the system with both SUDP and TCP traffic flows. In our study, we also find analytical results on the TCP traffic stable point are not accurate, so we derived new formulas on the TCP traffic stable point. We verified the proposed scheme with extensive NS2 simulations.

• Journal of Digital Convergence
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• v.13 no.11
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• pp.179-185
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• 2015

In this paper, we research the IPTV that is the convergence technique of TV and network technique and performed the performance evaluation of picture quality of IPTV in the situation of increasing the network traffic in the Internet environment. To do this, we constructed the mock Internet network similar to the real Internet environment and measured the quality of received video using V-Factor model according to the increase of network traffic, and analyzed the result of the experiment. Making use of the result of this paper for the threshold value of V-Factor, the measured factor of network performance, the measured factor of video performance in the watchable IPTV video quality.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.6
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• pp.2680-2696
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• 2018

The collaborative content caching system is an effective solution developed in recent years to reduce transmission delay and network traffic. In order to decrease the service end-to-end transmission delay for future 5G ultra-dense networks (UDN), this paper proposes a novel service migration method that can guarantee the continuity of service and simultaneously reduce the traffic flow in the network. In this paper, we propose a service migration optimization model that minimizes the cumulative transmission delay within the constraints of quality of service (QoS) guarantee and network condition. Subsequently, we propose an improved firefly algorithm to solve this optimization problem. Simulation results show that compared to traditional collaborative content caching schemes, the proposed algorithm can significantly decrease transmission delay and network traffic flow.