• Speech Sciences
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• v.11 no.4
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• pp.67-73
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• 2004

Packet loss, jitter, and delay in the Internet are caused mainly by the shortage of network bandwidth. It is due to queuing and routing process in the intermediate nodes of the packet network. In the Internet whose bandwidth is changing very rapidly in time depending on the number of users and data traffic, controlling the peak transmission bit-rate of a VoIP. system depending on the channel condition could be very helpful for making use of the available network bandwidth. Adapting packet size to the channel condition can reduce packet loss to improve the speech quality. It has been shown in [1] that a VoIP system with an AMR speech codec provides better speech quality than VoIP systems with fixed rate speech codecs. With the adaptive codec mode assignment. algorithm proposed in [1], in this paper, we performed the voice transmission experiments using the wireless LAN through the real Internet environment. Experimental results are analyzed and discussed with our findings.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.3B
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• pp.398-407
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• 2010

Due to increase of user requirement for various traffics and the advance of network technology, each distinct network has converge into FMC(Fixed Mobile Convergence) networks. However, we need to research the performance analysis of VoIP(Voice over Internet Protocol) in the FMC network to provide QoE for the voice user of FMC network. Therefore, this paper introduces the scenario which is the situation of voice quality degradation when a user uses VoIP to communicate with other users in the FMC network. Especially, this paper presents scenario in terms of the component of the network and finds the improvement point of voice quality. In the simulation results, three improvement points of voice quality are found as following: voice quality degradation by packet loss in the physical layer of the HSDPA network, by utilizing GGSN without QoS parameter mapping mechanism which is gateway between 3GPP and IP backbone, and by using non-QoS AP in the WLAN network.

• Journal of the Korea Society of Computer and Information
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• v.13 no.4
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• pp.161-167
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• 2008

A method to compute the size of packet and the optimal way to transmit the packets are proposed in this work for the VoIP communication using the MIL-STD-188-220C, military wireless Ad-hoc protocol which is used for the amicable communications of both speeches and data between several radiotelegraph. The expected time of data transmission is estimated beforehand, and then the size of package and transmission method are decided in the consideration of VoIP speech quality for the users as well as the data transmission quality of radiotelegraph.

• 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).

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.10a
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• pp.281-284
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• 2011

Smart phone is being used in the job as the ubiquitous society will Without being restricted by the time and place and devices. The rapid increase in the use of smart phones has brought the activation of the mobile job. And government agencies have brought in the transition to a smart society. In this paper, using a Voice over Internet protocol(VoIP) service for your smart phones to enhance security is the study of encryption technologies. External and internal signals, and call encryption and security standards of administrative agencies is the study of VoIP. Smart phone VoIP service is a study that security of equipment certificate, the internal signal and call encryption. This paper will contribute what using smart phone VoIP security and usability In smart generation.

• Proceedings of the Korean Information Science Society Conference
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• 2007.10a
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• pp.185-186
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• 2007

ITSP(Internet Telephony Service Provider)를 이용한 VoIP 서비스의 사용자가 증가함에 따라 VoIP 스팸은 큰 문제로 대두되고 있다. 기존의 일반 전화 때부터 사용되던 스팸은 실시간적 음성 통신이라는 특성상 콘텐츠 필터링을 하기 어렵기 때문에 롤 행위 패턴 조사를 통해 스패머(Spammer)를 구분하고 있다. 그러나 잘못된 오판으로 인한 문제와 스팸으로 인식하는 임계값을 넘지 않는 한도의 스팸 전송, 그리고 여러 사용자가 하나의 번호를 공유하여 사용하는 경우에는 여전히 스팸의 위협이 남아 있다. 이에 본 논문에서는 튜링 테스트를 이용한 VoIP 스팸 방지를 제안한다. 제안된 방법은 송신자에게 튜링 테스트를 거치게 하고, 튜링 테스트를 통과한 사용자만 수신자와 연결이 되는 방식으로 동작한다. 또한 튜링 테스트를 통과한 정상적인 사용자에게는 티켓을 줌으로써 재발신시 거쳐야하는 튜링 테스트의 번거로움을 줄일 수 있다. 제안된 방법은 ASUS WL-500G 무선 공유기 및 Asterisk IP-PBX에서 구현되었고 실험을 통해 제안된 방법의 유효성을 검증하였다.

• The Journal of the Korea institute of electronic communication sciences
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• v.9 no.6
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• pp.719-726
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• 2014

This paper addresses quality of experience(QoE) and how to measure and evaluate QoE including its subjective aspects. Adopting the real measurements on the field, a various quality metric have been measured for VoIP(voice over IP) service provided through a wireless interface of WiBro(Wireless Broadband). By analyzing the measured values and correlation between the metrics, we attempt to find a method to evaluate QoE of the VoIP service in a objective way. As a result, it has been shown that QoE of the VoIP service through WiBro network has close relation to the packet-level end-to-end delay, and the delay has close relation to received signal strength indicator(RSSI).

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.10a
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• pp.299-302
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• 2011

Recently, Voice over Internet protocol(VoIP) in IP-based wired and wireless voice, as well as by providing multimedia information transfer. Wired and wireless VoIP is easy on illegal eavesdropping of phone calls and VoIP call control signals on the network. In addition, service misuse attacks, denial of service attacks can be targeted as compared to traditional landline phones, there are several security vulnerabilities. In this paper, VoIP equipment in order to obtain information on the IP Phone is scanning. And check the password of IP Phone, and log in successful from the administrator's page. Then after reaching the page VoIP IP Phone Administrator Settings screen, phone number, port number, certification number, is changed. In addition, IP Phones that are registered in the administrator page of the call records check and personal information is the study of hacking.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.05a
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• pp.543-545
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• 2011

The concern which is abundant in MANET VoIP for comprising the mobility guarantee and mobile network is received without the infrastructure system between the mobile terminal node. However, because the access of system and border is easy, the issue which is big in the security problem becomes more than the wired network system with this convenience by the foreign network attacker differently. In this paper, we would like to the fundamental web network, NAT and concluding the security problem technology in which Firewall can inquire on MANET VoIP and whether it is appropriate or not which can solve this is proposed.

• Electronics and Telecommunications Trends
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• v.19 no.5 s.89
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• pp.114-124
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• 2004

유선 네트워크, 무선 이동통신 네트워크, 인터넷 등을 통합하는 유무선 통합 네트워크(BcN)에서는 VoIP기술을 사용하게 될 것이다. TTA 표준으로 2004년 7월에 제정된 VoIP 계층형 광대역 고품질 음성 코덱은 핵심계층에 G.711, G.723.1, G.729를 사용하므로 10종의 PT 를 설정하여 코덱을 협상한다. 이로 인하여 자기자신의 코덱 이외에도 G.711, G.723.1, G.729 등과 상호 호환이 되는 장점을 갖는다. 본 고는신규로 제정된 VoIP 계층형 광대역 고품질 음성 코덱을 네트워크에서 사용할 수 있도록 호 처리에 대한표준화를 추진하여야 하는데 이를 위한 표준 기술을 설명하고, 코덱과 호처리 관계 및 표준화 기술을 근거로 한 코덱 협상 처리 기술을 설명한다. 코덱 협상 처리 기술로서 PSTN/MSC 연동 코덱 협상 방안과All IP 코덱 협상 방안으로 구분하였다. All IP 코덱 협상 방안에서는 발신, 착신, MGC, 착신서버에서 호환성을 위한 호 처리 기능을 제공한다. 본 고의 호 처리 기술을 적용하면, VoIP 계층형 광대역 고품질 음성코덱은 기존 네트워크 장치 기능을 수정하지 않고 사용할 수 있다.