• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.3
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• pp.277-287
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• 1992

In this paper, we study the performance eualuations of the FDDI Network, by mathmeticlal analysis and simulation, in which the Integrated Voice/Data transmission system with voice traffic in synchronous mode and data traffic inasynchronous mode.For the mean waiting times of Voice/Data packet, we use two-state of Marcov models for voice traffic with talkspurt and silenci state, and the data traffic would traffic would transmit at the silence state of voice traffic. By the mean wating times, we analyze the relations between synchronous and asynchronous mode. As a result, using Sync/Async mode with voice and data, voice was not under influnece of data traffic. and in the same time,data can be tanaxmitted in a short waiting time, too.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.10
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• pp.2263-2268
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• 2015

This paper describes packet scheduling techniques that can be used to alleviate the impact of interference. The mechanism is consisted of interference estimation and master delay police. Proposed scheduling police is effective in reducing packet loss and delay. Another advantage worth mentioning, are the additional saving s in the transmitter power since packet are not transmitted when channel is bad. This paper gives that scheduling policy works only with data traffic since voice packets need to be sent at fixed intervals. However, if the delay variance is constant and the delay can be limited to a slot, it may be worthwhile to use DM packet for voice.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.9
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• pp.1304-1314
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• 1993

During the last several years, the demand of mobile communication is increasing rapidly due to the convenience of usage. Therefore, the evoluation scenario toward the new cellular network is needed. In this paper, the future prospect of cellular network is considered, and movable-boundary TDMA/BTMA protocol is proposed as a new voice/data integrated multiple access protocol for the future cellular packet radio networks. Then, the performance of movable-bounary TDMA/BTMA protocol is analyzed and compared with that of PRMA protocol by computer simulation. In the proposed movable-boundary TDMA/BTMA protocol, the voice traffic sensitive to delay time is served by TDMA protocol and the data traffic sensitive to loss is served by BTMA protocol. Also, the boundary of voice and data can be moved adaptively by usign SYN character, control byte, voice call counter, ect. Therefore. it could be seen that the performance of movable-boundary TDMA/BTMA protocol is better than that of PRMA protocol with respect to delay and throughput.

• Journal of Korea Multimedia Society
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• v.7 no.7
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• pp.896-904
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• 2004

In this paper, We present an efficient variable rate speech coder for spectral efficiency and packet processing technique for packet loss compensation of a voice codec with 10msec frame in VoIP service. Through disconnecting the users from the spectral resource during silence interval of about 60% period, a variable rate voice coder based on a voice activity detection(VAD) can increase spectral gain by two times. The performance of the method was analyzed by variation of detected voice activity factor and degraded speech frame ratio under various background noise level, and compared those of G.729B of ITU-T 8kbps standard speech codec. A method to compensate lost packets utilized addition of recovery data to a main stream and error concealment scheme for speech quality enhancement, the performance is verified by reconstructed speech quality. The proposed scheme can achieve spectral gain by two times or enhance speech quality by 3dB through reserved bandwidth of VAD. Therefore, the proposed method can enhance a spectral efficiency or speech quality of VoIP.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.10
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• pp.1994-2005
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• 2015

Voice over IP (VoIP) is a technology for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks. Instead of being transmitted over a circuit-switched network, however, the digital information is packetized, and transmission occurs in the form of IP packets over a packet-switched network which consist of several layers of computers. VoIP Service that used the various techniques has many advantages such as a voice Service, multimedia and additional service with cheap cost and so on. But the various frauds arises using VoIP because VoIP has the existing vulnerabilities at the Internet and based on complex technologies, which in turn, involve different components, protocols, and interfaces. According to research results, during in 2012, 46 % of fraud calls being made in VoIP. The revenue loss is considerable by fraud call. Among we will analyze for Toll Bypass Fraud by the SIM Box that occurs mainly on the international call, and propose the measures that can detect. Typically, proposed solutions to detect Toll Bypass fraud used DPI(Deep Packet Inspection) based on a variety of detection methods that using the Signature or statistical information, but Fraudster has used a number of countermeasures to avoid it as well. Particularly a Fraudster used countermeasure that encrypt VoIP Call Setup/Termination of SIP Signal or voice and both. This paper proposes the solution that is identifying equipment of Toll Bypass fraud using those countermeasures. Through feature of Voice traffic analysis, to detect involved equipment, and those behavior analysis to identifying SIM Box or Service Sever of VoIP Service Providers.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.3B
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• pp.161-167
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• 2007

In this paper, we evaluate the capacity of voice over internet protocol (VoIP) services over high-speed downlink packet access (HSDPA), in which frame-bundling (FB) is incorporated to reduce the effect of relatively large headers in the IP/UDP/RTP layers. Also, a modified proportional pair (PF) packet scheduler design supporting for VoIP service is provided. The main focus of this work is the effect of FB on system outage based on delay budget in radio access networks. Simulation results show that VoIP system performance with FB scheme is highly sensitive to delay budget. We also conclude that HSDPA is attractive for transmission of VoIP if compared to the circuit switched (CS) voice that is used in WCDMA (Release'99).

• The Transactions of the Korea Information Processing Society
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• v.7 no.4
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• pp.1211-1216
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• 2000

Asynchronous transfer mode (ATM) can be efficiently used to transport packet data services. The switching system will support voice and packet data services simultaneously from end to end applications. To guarantee quality of service (QoS) of the offered services, source rateot send packet data is needed to control the network overload condition. Most existing control algorithms are shown to provide the threshold-based feedback control technique. However, real-time voice calls can be dynamically connected and released during data services in the network. If the feedback control information delays, quality of the serviced voice can be degraded due to a time delay between source and destination in the high speed link. An adaptive algorithm based on the optimal least mean square error technique is presented for the predictive feedback control technique. The algorithm attempts to predict a future buffer size from weight (slope) adaptation of unknown functions, which are used fro feedback control. Simulation results are presented, which show the effectiveness of the algorithm.

• Journal of Internet Computing and Services
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• v.3 no.3
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• pp.67-74
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• 2002

Asynchronous transfer mode (ATM) can be efficiently used to transport packet data services. The switching system will support voice and packet data services simultaneously from end to end applications. To guarantee quality of service (QoS) of the offered services, source rate to send packet data is needed to control the network overload condition. Most existing control algorithms are shown to provide the threshold-based feedback control technique. However, real-time voice calls can be dynamically connected and released during data services in the network. If the feedback control information delays, quality of the serviced voice can be degraded due to a time delay between source and destination in the high speed link, An adaptive algorithm based on the optimal least mean square error technique is presented for the predictive feedback control technique. The algorithm attempts to predict a future buffer size from weight (slope) adaptation of unknown functions, which are used for feedback control. Simulation results are presented, which show the effectiveness of the algorithm.

• Proceedings of the KAIS Fall Conference
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• 2009.12a
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• pp.989-993
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• 2009

Wireless LAN(WLAN) is a rather mature communication technology connecting mobile terminals. IEEE 802.11 is a representative protocol among WLAN technologies. With the rising popularity of delay-sensitive real-time multimedia applications(video, voice and data) in IEEE 802.11 wireless LAN, it is important to study the MAC layer delay performance of WLANs. In this paper, performance for packet delay that recently have been proposed schemes is analysed in wireless LAN and proved performance results via simulation.

• Proceedings of the IEEK Conference
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• 2000.11a
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• pp.193-196
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• 2000

A new medium access control protocol is proposed to support stable wireless Internet services. Using the characteristics of the uplink Internet traffic and a Voice-Packet multi-session mode, the proposed protocol transmits the uplink Internet traffic via the voice traffic channel of silent duration in a multi-session mode. Out simulation results show that the proposed protocol guarantees stable wireless Internet services under heavy loads.