• 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.

• 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.

• 한국정보통신설비학회:학술대회논문집
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• 2008.08a
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• pp.97-101
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• 2008

Voice over IP is a telephone service which sends and receives the voices through the Internet. Because the infrastructure of wireless and mobile communication networks such as 3G, Wi-Fi and WiMAX has expanded, the study about Mobile VoIP, which provides the voice service from wireless network, has been actively in progress. Since Rei 6 HSPA in 3GPP and Rev A lxEVDO in 3GPP2, VoIP through the data channel is more efficient than circuit switch. It is predicted that VoIP over 4G will be more effective and 4G mobile VoIP business will be vitalized in the future. In addition, there are businesses which offer VoWLAN by using software such as Skype and Fring. They provide services which cheapen the price of international calls and long distance calls. This paper will present the Korean and other countries' mobile VoIP trends, its classification along the network connection, the study on techniques, and conditions of mobile VoIP. It also will be described a view of terminal convergence and service convergence.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.1
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• pp.24-51
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• 2011

When supporting both voice and TCP in a wireless multihop network, there are two conflicting goals: to protect the VoIP traffic, and to completely utilize the remaining capacity for TCP. We investigate the interaction between these two popular categories of traffic and find that conventional solution approaches, such as enhanced TCP variants, priority queues, bandwidth limitation, and traffic shaping do not always achieve the goals. TCP and VoIP traffic do not easily coexist because of TCP aggressiveness and data burstiness, and the (self-) interference nature of multihop traffic. We found that enhanced TCP variants fail to coexist with VoIP in the wireless multihop scenarios. Surprisingly, even priority schemes, including those built into the MAC such as RTS/CTS or 802.11e generally cannot protect voice, as they do not account for the interference outside communication range. We present VAGP (Voice Adaptive Gateway Pacer) - an adaptive bandwidth control algorithm at the access gateway that dynamically paces wired-to-wireless TCP data flows based on VoIP traffic status. VAGP continuously monitors the quality of VoIP flows at the gateway and controls the bandwidth used by TCP flows before entering the wireless multihop. To also maintain utilization and TCP performance, VAGP employs TCP specific mechanisms that suppress certain retransmissions across the wireless multihop. Compared to previous proposals for improving TCP over wireless multihop, we show that VAGP retains the end-to-end semantics of TCP, does not require modifications of endpoints, and works in a variety of conditions: different TCP variants, multiple flows, and internet delays, different patterns of interference, different multihop topologies, and different traffic patterns.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.05a
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• pp.414-418
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• 2003

VoIP is a core technology used to transmit both voice and data in an integrated packet' form. Within this technology, SIP is the signaling protocol used for 'real time' call services; particularly those where H323 is used. Yet, when considering the needs of mobile users, it is essential we integrate VoIp within the mobile technology so the mobile host is able to receive the 'packet' transported and by, and connected to, any available internet-address. For all this to occur, we need to improve Network Delay by reducing transmission problems associated with mobile services. If we are to obtain an optimal service then we must reduce any network delays which may arise from joining Mobile IP and VoIp services. This paper, therefore, considers how, unlike previous research, these delays may be improved through the use of the signaling technology\ulcorner SIP. It also considers how this research may be introduced into current wired and wireless integrated services enabling them to use the IP 'packet'.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.8B
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• pp.620-629
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• 2012

This paper proposes packet aggregation scheme for considering energy consumption of mobile devices on guaranteeing quality of voice. To our purpose, we analyze VoIP service on wireless multi-hop channel in terms of delay, packet loss and energy consumption when packet aggregation scheme is applied to VoIP service. Moreover, we induce a cost function with considering of tradeoff relation between quality of voice and energy consumption.

• Proceedings of the Korean Information Science Society Conference
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• 2002.04a
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• pp.163-165
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• 2002

현재 가장 중요하고 대중적인 통신수단으로 인식되고 있는 인터넷은 VoIP 기술을 이용하여 전화통신을 지원할 수 있게 되었다 VoIP 기술의 두드러진 장점은 기존의 인터넷망에 대해 추가적인 수정없이 음성전화 서비스를 통합함으로써 회선비용을 크게 절감하고, 단말의 이동성을 지원한다는 것이다. 이러한 장점으로 인해 VoIP는 향후의 네트워터에서 핵심 요소로 그 중요성이 점차 증가하고 있다. 본 논문에서는 다양한 VoIP 기술들 중 프로토콜 자체에 개인 이동성 지원이 명시되어 있고, 효과적인 이동호스트의 지원이 가능만 SIP을 기반으로 하여 단말이 이동중에 수신하는 beacon 신호의 감도와 빈도를 저장하여 예상 핸드오버 셀을 결정하는 SS buffer 메커니즘과 SIP server를 이용하여 계층적 이동성을 처리하는 hybrid mode를 갖춘 이동성지원 기법을 제안한다.

• International journal of advanced smart convergence
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• v.1 no.2
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• pp.59-63
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• 2012

The mVoIP service is the technology which focuses on the wireless data service as the IP network having the transmission rate of 100 Mbps classes on the high-speed middle of movement through the WiFi, WiBro and 3G mobile radio communication network, and etc. and is developed. Since 2010, the mVoIP (mobile VoIP) service shows the rapid growth due to 4G-LTE service seriously disclosed from July with the Smart phone and 2,011 it begins to be rapidly popularized. In this research, additionally the mVoIP service industry activation plan is presented with the trends of technology development including the chip-set/module/terminal, etc. based upon local and foreign market trend searchlight through the market demand analysis. The mVoIP service downloads App to the mobile apparatus and or can provide the service as the software through the WiFi network. Therefore, the change which is large in the products development aspect is to be have no. Expected to is being provided as the added service in the case of 4G-LTE as a matter of course, the service deployment where it is based on the market principle and demand needs is needed.

• Proceedings of the IEEK Conference
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• 2002.06a
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• pp.343-346
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• 2002

This paper presents a Prototype system developed for a Bluetooth interfaced VoIP system. The VoIP phone is developed based on tile implementation of a CELP coder on the TI 16bit DSP Processor A PC interfaced with Bluetooth module is used to designing a access point system. Host controller protocol stack is implemented to realize gateway between the wireless and wired line networks. A server application program for user management and call processing, which is based on TCP/IP peer to peer connection, is implemented for tile evaluation of overall interface system.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.57 no.11
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• pp.2124-2126
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• 2008

In the wireless communication systems, the demand of multimedia services is also increased. Unlike typical data packets, realtime service such as VoIP packets have delay bound and low loss rate requirement. In this paper we propose a new scheduling algorithm that be able to allocate resources to the different kinds of services such as VoIP and data packet. The proposed algorithm considers both time delay and channel condition toe determine the priority. Simulation results show that the proposed algorithm works more efficiently than the conventional algorithms.