• Information and Communications Magazine
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• v.21 no.7
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• pp.94-116
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• 2004

1968년 ARPA 네트워크의 출현 이후 음성 서비스 위주의 네트워크와 데이터 서비스 위주의 네트워크는 서로의 서비스를 수용하기 위해 노력해왔다. 음성 기반의 네트워크에서 데이터를 수용하기 위한 노력은 ‘꿈의 망’으로 그친 ISDN(Integrated Services Digital Network)으로 나타났고, 데이터 기반의 네트워크에서 음성을 수용하기 위한 노력은 보편적인 비연결형 데이터 서비스와는 대비되는 B-ISDN(Broadband-ISDN)으로 나타났다. 이후 다시 B-ISDN은 IP 서비스의 수용을 위해 ATM 교환기를 MPLS(Multi Protocol Label Switching) 교환기로 대치하여 보완하고 있지만, VoIP(Voice over IP)등 음성서비스의 제공에는 아직 완전한 해법이 제시되지 못하고 있다.(중략)

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.1
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• pp.32-39
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• 2015

The voice communications with wireless backbone based networks are evolving into a packet switching VoIP systems. In those networks, a call processing scheme is required for management of subscribers and connection between them. A VoIP service scheme for those systems requires reliable subscriber management and connection establishment schemes, but the conventional call processing schemes based on the centralized server has lack of reliability. Thus, the mutual-backup architecture of SIP-servers is required to ensure efficient subscriber management and reliable VoIP call processing capability, and the synchronization and call processing schemes should be changed as the architecture is changed. In this paper, a mutual-backup architecture of SIP-servers is proposed for wireless backbone based networks. A message format for synchronization and information exchange between SIP servers is also proposed in the paper. This paper also proposes a FSM scheme for the fast call processing in unreliable networks to detect multiple servers at a time. The performance analysis results show that the mutual backup server architecture increases the call processing success rates than conventional centralized server architecture. Also, the FSM scheme provides the smaller call processing times than conventional SIP, and the time is not increased although the number of SIP servers in the networks is increased.

• The KIPS Transactions:PartC
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• v.16C no.5
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• pp.661-668
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• 2009

The Extended real-time Polling Service (ErtPS) is proposed tosupport QoS of VoIP service with silence suppression which generates variable size data packets in IEEE 802.16e systems. If the silence is suppressed, VoIP should support Comfort Noise Generation (CNG) which generates comfort noise for receiver's auditory sense to notify the status of connection to the user. CNG mode in silent-period generates a data with lower bit rate at long packet transmission intervals in comparison with talk-spurt. Therefore, if the ErtPS, which is designed to support service flows that generate data packets on a periodic basis, is applied to silent-period, resources of the uplink are used inefficiently. In this paper, we proposed the Improved ErtPS algorithm for efficient resource utilization of the silent-period in VoIP traffic supporting CNG. In the proposed algorithm, the base station allocates bandwidth depending on the status of voice at the appropriate interval by havingthe user inform the changes of voice status. The Improved ErtPS utilizes the Cannel Quality Information Channel (CQICH) which is an uplink subchannel for delivering quality information of channel to the base station on a periodic basis in 802.16e systems. We evaluated the performance of proposed algorithm using OPNET simulator. We validated that proposed algorithm improves the bandwidth utilization of the uplink and packet transmission latency

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2018.10a
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• pp.568-570
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• 2018

Currently, with rapid progress and supply of mobile communication technology, LTE(Long Term Evolution) technology is expanded and widely used to industrial and emergency communications beyond earlier smart-phone based service. In this paper, transmission performance of voice traffic, one of railway communication service based on LTE-R as an application field of LTE technology, is analyzed. This study is performed performance analysis with level of application service and consider effects of satisfaction level for users. Computer Simulation based on ns(Network Simulation)-3 is used for analysis and VoIP(Voice over Internet Protocol) specification is used for voice traffics. Results of this paper is used to implement LTE-R networks and develope application services over LTE-R network.

• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.5
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• pp.528-533
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• 2010

Currently the telephone service using Internet grows and the recent introduction of a smart phone is expected to accelerate the trend. In particular, considering the domestic situation that the wireless broadband (WiBro) system deployed over the nation, the telephone service over WiBro can be a solution toward its fast expansion. Unlike telephone service over a conventional telephone network or mobile network, however, internet telephone cannot guarantee it service quality, which can be severer in a wireless environment such as a WiBro network. Therefore, a more strict and systematic management for controlling the quality of internet telephone service over WiBro in a more efficient way. As the first step to establish the management system, this paper proposes a scheme to manage the quality of internet telephone service over WiBro and introduces a software developed for the purpose. The developed software is installed on a user terminal and facilitates efficient service quality management by measuring the quality of internet telephone service over WiBro in terms of VoIP metric, network metric, and wireless metric.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.9
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• pp.373-383
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• 2003

In this paper we discuss the effect of the delay limit and the packet size related to the quality of service on a VoIP system using the Internet. We also provide a guideline to determining the optimal packet size of the voice data for a given delay limit. Empirical studies are done with two personal computers connected through the packet switched public IP network. The sender encodes the voice signal from the microphone to get PCM and ADPCM data and sends the data to the receiver using UDP packets. The receiver plays the reconstructed voice from the stream with lost and delayed packets. The quality of the reconstructed voice is evaluated offline by the MNB (Measuring Normal Block) method using the data acquired from the both sides. The result shows that under the delay limit of 100ms for 40Kbps, 32Kbps and l6Kbps of ADPCM data, the minimum packet size should be 300bytes, 400bytes and 600bytes respectively and the maximum packet size should be l200bytes commonly for the best quality of voice.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.16 no.7
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• pp.4878-4883
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• 2015

The IP Multimedia Subsystem(IMS) is a framework that provides access to the content of Internet and Telecom services anytime and anywhere with guaranteed Quality of Service(QoS) and manageability by separating control functions from bearer and services. The Service Delivery Platform(SDP) provides common interfaces and protocols to deploy existing or new services in an efficient way. Therefore SDP over IMS plays a role of bridge between established network and new IMS network by simplifying the interaction among application services. In order to enrich the multimedia network communication, we try to deploy the Push-to-talk over Cellular(PoC) service which is considered as the outstanding and distinguished half-duplex Voice over IP(VoIP) application service among deployable candidate services over mobile network. In this paper we investigate the advantages of PoC service and PoC architecture firstly, and then focus on the its practical implementation for the prototype to validate the feasibility of its deployment and realization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8B
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• pp.797-804
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• 2009

Recently, SIP(Session Initiation Protocol) is used to set up and manage sessions for multimedia applications such as VoIP(Voice over IP) and IMS(IP Multimedia Subsystem). However, because SIP operates over the Internet, it is exposed to pre-existed internet security threats such as service degradation or service disruptions. Multimedia applications which are delay sensitive even suffers more from the threats mentioned above. The proposed methods so far to detect SIP INVITE flooding are CUSUM(Cumulative Sum), Hellinger distance and adaptive threshold, but among methods only take normal state into consideration. So, it is not capable of adapting the condition of the network congestion which are dynamically changing. In this paper, SIP INVITE flooding detection algorithm considering network congestion which enables efficient detections of such attacks is proposed. The proposed algorithm is expected to detect other types of attacks such as BYE and CANCEL more precisely compared to other methods.

• Annual Conference of KIPS
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• 2013.11a
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• pp.793-796
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• 2013

스마트폰, IPTV(Internet Protocol Television), VoIP(Voice over Internet Protocol) 등의 인터넷을 활용한 서비스의 증가는 IPv4(Internet Protocol Version4)의 주소 부족문제를 야기시켰으며 이를 해결하기 위한 장기적인 해결방안으로는 IPv6(Internet Protocol Version6)가 제시되었고 단기적으로는 NAT(Network Address Translator)가 제안되었다. NAT는 사설 IP 주소를 공인 IP 주소로 활용하여 네트워크에 접속할 수 있도록 지원하며 주소 부족 문제를 해결하고 내부 네트워크를 보호하는 긍정적인 기능도 하지만 역으로 해커들에게 숨은 공간을 제공하는 역할을 하기도 한다. 본 논문에서는 NAT 기능을 활용한 IP 공유기를 통해 내부 보안 프로세스를 우회할 수 있는 단말기를 검출하는 탐색 알고리즘을 분석하고 이를 병렬화하여 정확도를 높일 수 있는 검출 시스템을 연구하고자 한다.

• Proceedings of the KSPS conference
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• 2003.10a
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• pp.109-112
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• 2003

In real time packetized voice applications, missing frames is a major source of voice quality degradation. Thus packet loss concealment(PLC) algorithms are needed to guarantee the QoS of the VoIP. Still current speech codecs for VoIP work poor when consecutive packet losses are issued. In this paper, we proposed a new PLC algorithm for the G.729 codec. Our algorithm works better especially when the consecutive packet loss occurs mainly because it adopts an adaptive gain controller utilizing the number of missing packet information combined with a fixed codebook vector estimation algorithm and LPC bandwidth expansion.