• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.12
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• pp.18-26
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• 2003

In this paper, we design and implement a QoS control function in the SIP-based VoIP system. As a network infrastructure for VoIP service, we select the Intserv over Diffserv architecture where the network resources are managed by a call admission control mechanism. The SIP protocol extended to support QoS signaling procedure is modulized to operate independently with the infrastructure. The performance of the QoS-enabled VoIP system is verified by experiments.

• Proceedings of the Korea Society of Information Technology Applications Conference
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• 2002.06a
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• pp.51-52
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• 2002

• Journal of the Korea Convergence Society
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• v.3 no.3
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• pp.7-12
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• 2012

mVoIP (mobile Voice over Internet Protocol) service is a technology to transmit voice data through an IP network using mobile device. mVoIP provides various supplementary services with low communication cost. It can maximize the availability and efficiency by using IP-based network resources. In addition, the users can use voice call service at any time and in any place, as long as they can access the Internet on mobile device easily. However, SIP on mobile device is exposed to IP-based attacks and threats. Observed cyber threats to SIP services include wiretapping, denial of service, and service misuse, VoIP spam which are also applicable to existing IP-based networks. These attacks are also applicable to SIP and continuously cause problems. In this study, we analysis the threat and vulnerability on mVoIP service and propose several possible attack scenarios on existing mobile VoIP devices. Based on a proposed analysis and vulnerability test mechanism, we can construct more enhanced SIP security mechanism and stable mobile VoIP service framework after eliminating its vulnerability on mobile telephony system.

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

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.10a
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• pp.277-280
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• 2011

VoIP is a technology of voice communication, using the existing internet network which sends and receives voice packets. VoIP has an advantage that VoIP is cheaper than an existing telephony, and the tech is vitalized lately. But recently you can download Volp Application in the Market that have a vulnerability(Anyone Can Upload). This weakness is wrongfully used that People are downloaded by encouraging about malignant code is planted. Signal intercepts indicates from this case. and paralysis by DDoS Attack, bypass are charged for hacking. Judging from, security threat of VolP analysis and take countermeasures. In the thesis we analyze the VoIP security caused on 'Soft Phone' and 'Smart Phone', and figure out security policies and delineate those policies on the paper.

• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.3
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• pp.275-280
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• 2010

In this paper, VoIP(Voice over Internet Protocol) transmission performance for MANET(Mobile Ad-hoc Networks) is improved and analyzed with packet aggregation scheme which is aggregating some of short length packets to one large packet and sending to networks. VoIP simulator based on NS(Network Simulator)-2 is implemented and used to measure performance of VoIP traffic transmission. In this simulation, VoIP traffics are generated with parameters of some codes such as G.711, G.729A, GSM.AMR and iBLC. MOS(Mean Opinion Score), end-to-end network delay, packet loss rate and transmission bandwidth are measured. Performance improvements of 98% for MOS, 6.4times for end-to-end network delay, 32times for packet loss rate is shown as simulation results. On the other hand, transmission bandwidth is increased about maximum 10%. Finally, VoIP implementation guide for the performance with packet aggregation is suggested.

• The Journal of the Korea institute of electronic communication sciences
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• v.6 no.1
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• pp.49-54
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• 2011

In this paper, End-to-End VoIP(Voice over Internet Protocol) performance over large scale MANET(Mobile Ad-hoc Networks) is evaluated. VoIP implementation model, which can be used for large scale MANET, is suggested through this evaluation. VoIP simulator based on NS-2 network simulator is used for performance measurement. Call connection ratio as a connection performance, MOS and network delay as a transmission performance, packet loss rate as other performance is measured with this simulation. Finally, maximum $10{\sim}15km^2$ as a single MANET, 500 nodes and 100 call connections is suggested for VoIP implementation and operation conditions over large scale MANETs.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.11a
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• pp.839-843
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• 2002

In this paper, we present a next generation VoIP network architecture. Specifically, we present key components such as SoftSwitch and media gateway, and important protocols for VoIP services. We also present basic components and mechanisms for VoIP QoS. The architecture presented in this paper is able to support open interfaces and multimedia traffic, and therefore various multi-services can be supported efficiently using the architecture.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.990-994
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• 2003

This paper describes the VoIP(Voice over Internet Protocol) SoC(System on Chip) Design and the test board development for the function verification to support voice communication services using Internet. To implement the simple system of configuration, we designed the VoIP SoC which have ARM922T of 32bit microprocessor, IP network interface, voice signal interface, various user interface function. Also we developed test program and communication protocol to verify the function of this chip. We used several tools of design and simulation, developed and tested a test board with Excalibur which includes ARM922T microprocessor and FPGA.

• Journal of Internet Computing and Services
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• v.10 no.6
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• pp.27-36
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• 2009

PSTN (public switched telephone network) provides voice communication service, whereas IP network provides data oriented service, and we can use IP network for multimedia transport service (e.g. voice over IP service) with economic price. In this paper, we propose RoIP (robot on IP) service scenario, signaling call flow, and implementation to provide home automation and monitoring service for remote site users. In our scheme, we used a extended SIP (session initiation protocol) for signaling protocol between remote site users and home robots. For our bearer transport control, we implemented H.263 video codec over RTP (real-time transport protocol) and additionally DTMF (dual tone multi-frequency) transport for robot actuator control. We implemented our scheme on home robots and experimented with KTF operator network, and it shows good communication quality (average MOS = 9.15) and flexible robot controls.