• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.5
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• pp.967-973
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• 2009

Recently, as 3rd-generation mobile communication services using high-speed data rate system are widely employed, the demand for a variety of real-time data services such as VoIP service are also increased. Unlike typical data packets, VoIP packets have delay bound and low loss rate requirement. In this paper we propose a new scheduling algorithm that schedule two deferent kinds of packets efficiently, considering the characteristics of VoIP. Basically this algorithm considers both time delay and channel condition and gives priority depending on the time delay. Simulation results show that the proposed algorithm works more efficiently than conventional algorithms.

• 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.

• The Journal of the Korea Contents Association
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• v.2 no.3
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• pp.119-126
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• 2002

The QoS(Quality of Service) guarantee mechanism is one of critical issues in the wireless network. Real-time applications like VoIP(Voice over IP) in All-IP networks need smooth handoffs in order to minimize or eliminate packet loss as a Mobile Host(MH) transitions between network links. In this paper, we design a new multicasting algorithm using DB(Dynamic Buffering) mechanism for Mobile-IPv6. A key feature of the new protocol is the concepts of the DB and MRA(Multicast Routing Agent) to reduce delivery path length of the multicast datagram. Particularly, the number of tunneling and average routing length of datagram are reduced relatively, the multicast traffic load is also decreased.

• 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.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.33 no.2
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• pp.133-138
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• 2014

Packet arrival-delay variation, so-called 'jitter' is one of the main factors that degrade the quality of voice in mobile devices at the Voice over Internet Protocol (VoIP). To resolve this issue, a playout scheduling based on adaptive jitter estimation for enhancing VoIP speech quality is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. The experimental results have shown that the proposed algorithm delivers high voice quality in unstable network environment.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.10
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• pp.2462-2472
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• 2014

Recently, the computing paradigm has been changing and VoIP technology is being revisited to support various services. In this paper, we have designed and implemented the system of software PBX open source Asterisk, hardware platform, and mobile devices to support secured voice service based on VoIP. Specially, we designed the various platform from single board to servers based on XenServer in hardware platform. And we verified the delay test of network traffics and the secured voice communication test based on this platform.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2010.10a
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• pp.414-416
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• 2010

WiBro network technologies developed mainly by Korea are one of very promising 4G technologies. WiBro has been standardized as international standard called Mobile WiMAX, and has been deployed in many countries. Emergency Services are infrastructural servicesa and very essential to induce subscribers from other types of networks. This paper propose a network architecture for supporting emergency services for mobile VoIP services in WiBro networks. The proposed architecture is based on WMF's architecture for internationa compatibilities, and reflects national considerations on interfacing PSAP for domestic compatibilities.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.1
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• pp.145-152
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• 2009

IEEE 802.16e standard, a kind of WMAN standard, was established to support data services with cheaper cost to mobile users than traditional mobile communications system and wireless LAN system can do. In this paper, we propose an uplink QoS packet-scheduler for VoIP service which can be installed in IEEE 802.16 system and evaluate its performance with NS-2 network simulator. The proposed QoS packet-scheduler consists of three procedures: determining scheduler interval, determining the amount of resource assignment, and deciding which mobile station the base station should serve first among multiple mobile stations. According to numerical results, the proposed QoS packet-scheduler can provide more increased system capacity by 220% than UGS service scheme does and by 25 % than ertPS service scheme does.

• The Journal of the Korea institute of electronic communication sciences
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• v.6 no.1
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• pp.55-60
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• 2011

The mobile internet telephony service rapidly grows according to the extending deployment of smartphones. Unlike telephony service over a conventional public switched telephone network (PSTN) or mobile network, internet telephony service cannot guarantee its service quality, which can be severer in a wireless environment. Therefore, a more strict and systematic quality management is required for successful settlement and popularization of mobile internet telephony service. Existing quality management scheme using a specific measurement equipment cannot measure all the time so that it performs late management. In order to overcome the problem, this paper develops a software that can be equipped on a user terminal and measures the service quality all the time. By using the developed software, all-time and user-activating service quality monitoring can be supported.