• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.5 no.7
• /
• pp.1272-1279
• /
• 2001

Owing to the change of various services requirement and the change of communication paradime as the development of new transit network technology: ATM, MPLS and DWDM, the latest PSTN is evolving into the next generation network, which have the very high speed integrated packet network of open-type structure. but in the next generation network, the existing voice service is being researched for the continuous technology development as the important service from the viewpoint of communication business. This paper suggests the method of efficient bandwidth management in the media gateway that can accommodate the various new services as well as the existing voice services by appling to the next generation network.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.11 no.3
• /
• pp.309-316
• /
• 2016

Recently, the voice service over a mobile communication system tends to be provided based on the packet-based technology. Even though the sufficient transmission rate is supported by LTE mobile communication system, the quality of VoIP service that is experienced by the user can be degraded by the change in the transmission conditions and the terminal mobility. This paper has established an environment on which experiments are conducted for the different values of the major parameters that represent the transmission conditions. The result can contribute to the decision of the requirement that the mobile system should meet for maintaining the quality of VoIP service.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.14 no.5
• /
• pp.1109-1116
• /
• 2010

In this paper, some performance characteristics of VoIP(Voice over Internet Protocol) for MANET(Mobile Ad-hoc Networks) with simulation is studied and appropriate condition for implementation of VoIP service is suggested. VoIP simulator is implemented with NS(Network Simulator)-2. VoIP traffic for simulation is generated with some codecs of G.711, G.723.1, G.726-32, G.729A, GSM.AMR and iLBC. As simulation results for traffic transmission under $670{\times}670m$ 50node MANET environment, performance data for MOS(Mean Opinion Score), network delay, packet loss rate and transmission bandwidth are measured. Normalized analysis about measured results shows that maximum VoIP connection satisfying VoIP service quality condition is 15.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2009.10a
• /
• pp.795-797
• /
• 2009

Voice over Internet Protocol(VoIP) service becomes more and more popular nowadays. As such, it is developed over many kinds of network models, especially wireless networks. Mean Opinion Score(MOS) computes the QoS of VoIP service which should be supported by robust network environment. However, MANET is not stable enough to supply high MOS values for VoIP service. In this paper, VoIP service over MANET is simulated using ns-2(Network Simulation 2). In oder to get different MOS values in the results, we differentiate between network environments by adjusting the parameters of MANET.Through comparing the results we can know how to improve the QoS.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• v.9 no.1
• /
• pp.625-629
• /
• 2005

According to the rapid development of the information and communication technology, the demand on the efficient compression technology for the multimedia data is increased magnificently. In this Paper, we designed new compression algorithm structure using wavelet base for the compression of ECG signal and audible signal data. We examined the efficiency of the compression between 2-band structure and wavelet packet structure, and investigated the efficiency and reconstruction error by wavelet base function using Daubechies wavelet coefficient and Coiflet coefficient for each structure. Finally, data were compressed further more using Huffman code, and resultant Compression Rate(CR) and Percent Root Mean Square difference(PRD) were compared with those of existent DCT.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.28 no.8B
• /
• pp.707-715
• /
• 2003

In this paper, we propose a new MAC(Medium Access Control) protocol using Movable-boundary, which tries to guarantee Qos for multimedia traffic in the slotted CDMA system. In this scheme, the traffic scheduler assigns channel resource according to the packet priority per service class and adapts the Movable-boundary concept in which the minimum resource is assigned to each traffic class; the remaining resource if it is available can be assigned dynamically according to the temporal demand of other traffic classes. For performance analysis, we performed computer simulations to obtain throughput and packet loss rate and compared the results with Fixed-boundary system. We observed that the error rate of voice traffic could be maintained below a prescribed value while bursty traffic such as video source shares the same channel. In comparison with Fixed-boundary scheme, our protocol exhibits better throughput and packet loss rate performance.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.14 no.8
• /
• pp.1783-1791
• /
• 2010

Mobile WiMAX system provides broadband wireless access with a variety of services such as voice, video and data communications and providing QoS for each of these services become an important issue. In Mobile WiMAX system, it is important to allocate resources appropriately in order to support the efficient utilization of resources among various real-time and non real-time services. Although many packet scheduling schemes for real-time services in OFDMA system have been proposed, it need to be modified to apply to Mobile WiMAX system. Since Mobile WiMAX supports five types of service classes, QoS constraints of each class should be taken into consideration. In this paper, we propose an efficient packet scheduling scheme to support various services by considering the QoS constraints of each class.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.34 no.6A
• /
• pp.458-467
• /
• 2009

Long Tenn Evolution (LTE) is the next generation mobile phone technology which has being standardized by the Third Generation Partnership Project (3GPP). In the existing mobile communication networks, voice traffic is delivered through circuit switched networks. In LTE, however, all kinds of traffic are transferred through IP based packet switched networks which has best-effort characteristic. Therefore, providing QoS in LTE system is difficult. In order to provide QoS in LTE, RRM is very important. Especially, in part of RRM, call admission control (CAC) performs an important function to reduce network congestion and guarantee a certain level of QoS for on-going calls. In this paper, we propose a CAC algorithm in order to provide QoS for various kinds of services in LTE system. The performance of the proposed algorithm is evaluated with various simulation environments. The results show that the proposed algorithm provides QoS through rejections of requested calls. Especially, the proposed CAC algorithm can be satisfied with packet delay requirement defined in LTE specification.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
• /
• v.14 no.4
• /
• pp.247-256
• /
• 2021

In this paper, we propose a packet loss concealment (PLC) algorithm using pitch harmonic motion prediction and adaptive signal amplitude prediction and. The spectral motion prediction method divides the spectral motion of the previous usable frame into predetermined sub-bands to predict and restore the motion of the lost signal. In the proposed algorithm, the speech signal is classified into voiced and unvoiced sounds. In the case of voiced sounds, it is further divided into pitch harmonics using the pitch frequency to predict and restore the pitch harmonic motion of the lost frame, and for the unvoiced sound, the lost frame is restored using the spectral motion prediction method. When the continuous loss of speech frames occurs, a method of adjusting the gain using the least mean square (LMS) predictor is proposed. The performance of the proposed algorithm was evaluated through the objective evaluation method, PESQ (Perceptual Evaluation of Speech Quality) and was showed MOS 0.1 improvement over the conventional method.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.25 no.4A
• /
• pp.451-463
• /
• 2000

In this paper, we design the radio data link protocols with QoS provisioning for mobile multimedia such as voice, data, and video in CDMA-based ATM networks, and analyze the performance of the data link protocols. To support mobile multimedia traffic, the required QoS parameters and the characteristics are analyzed, and wireless protocol stacks are proposed for integrating the wireless access network and ATM transport networks, and radio data link protocols are designed for provisioning QoS Control. The data link protocols are analyzed assuming that the system is supporting voice and data traffic simultaneously. In case of data traffic, the delay and throughput of SREJ ARQ and Type-1 Hybrid ARQ scheme are compared, and in case of voice traffic, the packet loss rate of BCH coding is analyzed according to the varying data traffic loads. The results indicate that the adaptive radio link protocols are efficient to support QoS requirements while the complexities are increased.