• Journal of KIISE:Computer Systems and Theory
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• v.26 no.4
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• pp.500-512
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• 1999

본 연구에서는 CDMA 기술에 기반한 IMT2000 시스템을 연구중인 cdmaOne 3세대 CDMA MAC을 이용한 무선 패킷 음성 서비스 프로토콜(WPVCP : Wireless Packet Voice Convergence Protocol)을 설계하고 성능을 평가하였다. 설계한 WPVCP 프로토콜은 회선형 서비스 중심인 이동 통신의 음성 서비스 구조를 패킷형 서비스 개념에서 지원하도록 하였으며, 이를 통하여 제한된 무선자원의 활용을 극대화하였다. 아울러, 묵음 구간으로 인한 기존 회선형 방안의 음성 서비스와 제안한 패킷형 음성 서비스를 비교 분석한 결과 제안한 방안의 성능이 end-to-end 의 음성 품질을 지원하는 구조에서 최대 250% ~280% 수준의 가입자 증대효과를 나타내는 것으로 확인하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.1C
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• pp.48-58
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• 2002

As internet traffic increases, the problem that it should be efficiently accepted in to the existing voice network in the pending problem importantly to the existing communication corporations. The feature of next generation exchange network is made up of the form of integration network that connect data traffic(internet service. Etc) with the existing voice network and it can be showed with very diverse aspects according to the constitution time of network or the characteristics of business. The progress strategy that develop the existing circuit based communication network into packet-based communication network can be divided into two in a large scale according to the application position These are VoDSL application method(Technology that packetize access network) and softswitch technology application method(after packetizing relay network, packetizing that into the access network). In this paper, we will deduce the desirable technology that can construct packet-based next generation exchange networks in the structure of the existing communication network environment. We will perform the research on a device to offer the necessary core technique VoSDL service with realistic resolutions primarily.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.11b
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• pp.1301-1304
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• 2002

통신 환경이 변하면서, 기존의 서비스에 따라 크게 음성망과 패킷망으로 구분되던 망 구조가 하나의 통합된 망에서 모든 서비스를 제공하는 구조로 진화하고 있다. 그리고, 이러한 서비스를 가능하게 하는 기술로서 VoIP(Voice over IP)가 최근까지도 계속 각광받고 있다. 그러나, 많은 노력에도 불구하고, 음성서비스와 같은 실시간 서비스의 엄격한 품질 요구조건을 보장하는 문제 때문에 VoIP 기술의 실제 적용이 지연되고 있다. 이에 본 논문에서는 통합망의 패킷 전달망을 ACE2000 MPLS 시스템 기반의 MPLS 망으로 구축함으로써 음성서비스의 품질을 보장하는 망 구조를 제시하고자 한다. 아울러 TE Server를 이용해서, 음성호를 전달하는 ER-LSP(Explicit Routed Label Switched Path)를 설정하는 호 설정 절차를 제시하였다.

• 한국IT서비스학회:학술대회논문집
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• 2009.05a
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• pp.365-368
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• 2009

고속 인터넷 망 구축과 멀티미디어 통신 수요의 증가에 따라 VoIP는 기존의 PSTN 망의 대체 혹은 확장 기술로서 지속적으로 검증되어 왔다. 음성 데이터 처리 규약들 중 SIP는 다른 규약에 비해 신호 처리 단계가 간단하기 때문에 이를 기반으로 RTP를 활용하여 음성 통신 시스템을 구축하는 사례가 늘어나고 있다. 그러나 RTP의 특성상 패킷을 처리할 때마다 복원 과정이 필요하며, 다중 세션으로 통신이 발생할 경우 전체 패킷들의 관리가 복잡해지므로 이들 간에 혼선 없이 데이터를 처리 및 유지할 수 있는 방법론이 요구된다. 본 논문에서는 SIP 기반의 IP 전화를 통해서 고객과 상담원 간의 통화 이벤트가 발생하는 일반 콜센터 환경에서 RTP 음성 데이터를 처리하는 다중 세션 어플리케이션의 구축 사례를 제시한다. 구현한 시스템은 IP 전화에서 발생하는 통화 내역을 통합 스위치 서버에서 포트 미러링하여 녹취 및 녹음 서버로 전송하며, 전송된 패킷 정보들의 세션이 유지되고 있는 동안 음성 데이터를 실시간으로 모니터링한다.

• The KIPS Transactions:PartC
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• v.19C no.2
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• pp.119-126
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• 2012

This paper covers the service quality of packet-based voice service that is provided over wireless broadband (WiBro) and high speed downlink packet access (HSDPA) systems. Using a measurement software that has been developed in the course of preparing a advanced service quality management scheme for the packet-based voice service over wireless networks, a huge scale of experiment is conducted to measure the real quality of the voice service. Based on our analysis of the measurement results, the service quality of the voice service is supposed to be quite good over both wireless systems. In addition, another experiment to investigate the effect of degradation of wireless transmission conditions on the service quality of the voice service shows the values of wireless service metrics in which mean opinion score (MOS) starts to decrease.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.5
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• pp.916-922
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• 2001

A medium access control protocol is proposed for integrated voice and data services in the packet CDMA network with a small coverage. Uplink channels are composed of time slots and multiple spreading codes for each slot. This protocol gives higher access priority to the delay-sensitive voice traffic than to the data traffic. During a talkspurt, voice terminals reserve a spreading code to transmit multiple voice packets. On the other hand, whenever generating a data packet, data terminals transmit a packet based on the status Information of spreading codes in the current slot, which is received from base station. In this protocol, voice packet does not come into collision with data packet. Therefore, this protocol can increase the maximum number of voice terminals.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.1
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• pp.20-33
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• 2002

Usage of ATM AAL2 packets becomes dominant to increase transmission efficiency of voice traffic in the backbone network. In case of voice service that uses AAL2 mechanism, if resources of network are enough, connection of new call is accepted. However, due to packets generated by the new call, transmission delay of packets from old calls can increase sharply. To control this behavior, in this paper we present an AAL2 buffer management scheme that allocates a virtual buffer to each call and after calculating its propagation delay variation(PDV), decides to drop packets coming from each call according to the PDV value. We show that this packet dropping algorithm can effectively prevent abrupt QoS degradation of old calls. To do this, we analyze AAL2 packet composition process to find a critical factor in the process that influences the end-to-end delay behavior and model the process by K-policy M/D/1 queueing system and MIN(K, Tc)-policy M/D/1 queueing system. From the mathematical model, we derive the probability generating function of AAL2 packets in the buffer and mean waiting time of packets in the AAL2 buffer. Analytical results show that the AAL2 packet dropping algorithm can provide stable AAL2 packetization delay and ATM cell generation time even if the number of voice sources increases dramatically. Finally we compare the analytical result to simulation data obtained by using the COMNET Ⅲ package.

• Journal of KIISE:Information Networking
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• v.29 no.2
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• pp.141-155
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• 2002

A medium access control protocol is proposed for integrated voice and data services in the packet CDMA network with a small coverage. Uplink channels are composed of time slots and multiple spreading codes for each slot. This protocol gives higher access priority to the delay-sensitive voice traffic than to the data traffic. During a talkspurt, voice terminals reserve a spreading code to transmit multiple voice packets. On the other hand, whenever generating a data packet, data terminals transmit a packet based on the status information of spreading codes in the current slot, which is received from base station. In this protocol, voice packet does not come into collision with data packet. Therefore, this protocol can increase the maximum number of voice terminals.

• The Journal of the Korea institute of electronic communication sciences
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• v.6 no.6
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• pp.823-829
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• 2011

This paper covers the service quality of packet-based voice service that is provided over a wireless broadband (WiBro) network. Using a measurement software that has been developed in the course of preparing a advanced service quality management scheme for the packet-based voice service over a wireless network[2][3], a huge scale of experiment is conducted to measure the real quality of the voice service. Based on our analysis of the measurement result, the service quality of the voice service is supposed to be quite good over WiBro networks. In addition, another experiment to investigate the effect of degradation of wireless transmission conditions on the service quality of the voice service shows the values of wireless service metris in which mean opinion score (MOS) starts to decrease.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.2
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• pp.249-255
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• 2012

This paper covers the service quality of packet-based voice service that is provided over a high speed downlink packet access (HSDPA) system. Using the measurement software that has been developed in the course of preparing a advanced service quality management scheme for the packet-based voice service over a wireless network[2][3], a huge scale of experiment is conducted to measure the real quality of the voice service. Based on our analysis of the measurement result, the service quality of the voice service is supposed to be quite good over HSDPA system. In addition, another experiment to investigate the effect of degradation of wireless transmission conditions on the service quality of the voice service shows the values of wireless service metrics in which mean opinion score (MOS) starts to decrease.