• Journal of the Institute of Convergence Signal Processing
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• v.10 no.3
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• pp.181-185
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• 2009

Recently, much attention to a self-organizing mobile ad-hoc network is escalating along with progressive deployment of wireless networks in our everyday life. Being readily deployable, the MANET (mobile ad hoc network) can find its applications to emergency medical service, customized calling service, group-based communications, and military purposes. In this paper we investigate a time slot exchange problem found in the time slot based MAC, that is designed for IEEE 802.11b interfaces composing a MANET. The paper provides a method to maintain the quality of voice call by providing a new time slot when the channel assigned for that time slot gets noisy with interferences induced from other nodes, which belong to the same and/or other subgroups. In order to assess the performance of the proposed algorithm, a set of simulations using the OPNET modeler has been performed assuming that the IEEE 802.11b interfaces are operating under a modified MAC, which is a time slot based reservation MAC implemented in the PCF part of the superframe. In a real-time voice call service over a MANET of a size 500 ${\times}$ 500 meter squares with the number of nodes up to 100, the simulation results are collected and analyzed with respect to the packet loss rate and packet delay. The results show us that the proposed time slot exchange protocol improves the quality of voice call over that of plain DCF.

• The KIPS Transactions:PartC
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• v.17C no.3
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• pp.299-306
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• 2010

IPTV and VoIP services are considered as killer applications over Mobile WiMAX network, which provideshigh mobility and data rate. Among those which affect the quality of voice in those services, the jitter buffer or playout buffer can compensate the poor voice quality caused by the packet drop due to frequent route change or differences among routes between service endpoints. In this paper, we analyze the correlation between the sleep interval length and jitter buffer size in order to guarantee a predefined level of voice quality. For this purpose, we present an end-to-end delay model considering additional delay incurred by the WiMAX PSC-II sleep mode and a VoIP service quality requirement based on the delay constraints. Through extensive simulation experiments, we also show that the increase of jitter buffer size may degrade the voice quality since it can introduce additional packet drop in the jitter buffer under WiMAX power saving mode.

• Journal of Internet Computing and Services
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• v.3 no.3
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• pp.67-74
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• 2002

Asynchronous transfer mode (ATM) can be efficiently used to transport packet data services. The switching system will support voice and packet data services simultaneously from end to end applications. To guarantee quality of service (QoS) of the offered services, source rate to send packet data is needed to control the network overload condition. Most existing control algorithms are shown to provide the threshold-based feedback control technique. However, real-time voice calls can be dynamically connected and released during data services in the network. If the feedback control information delays, quality of the serviced voice can be degraded due to a time delay between source and destination in the high speed link, An adaptive algorithm based on the optimal least mean square error technique is presented for the predictive feedback control technique. The algorithm attempts to predict a future buffer size from weight (slope) adaptation of unknown functions, which are used for feedback control. Simulation results are presented, which show the effectiveness of the algorithm.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.3 no.3
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• pp.251-265
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• 2009

VoIP service is the transmission of voice data using SIP protocol on an IP-based network. The SIP protocol has many advantages, such as providing IP-based voice communication and multimedia service with low communication cost. Therefore, the SIP protocol disseminated quickly. However, SIP protocol exposes new forms of vulnerabilities to malicious attacks, such as message flooding attack. It also incurs threats from many existing vulnerabilities as occurs for IP-based protocol. In this paper, we propose a new virtual proxy to cooperate with the existing Proxy Server to provide state monitoring and detect SIP message flooding attack with IP/MAC authentication. Based on a proposed virtual proxy, the proposed system enhances SIP attack detection performance with minimal latency of SIP packet transmission.

• The Transactions of the Korea Information Processing Society
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• v.6 no.11S
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• pp.3190-3198
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• 1999

Because of the rapid growth of the mobile communication, the need for the mobile internet access has grown up as well. since the current mobile communication network, however, is optimized for a voice communication system, which exclusively occupies a channel for a given time, it is not suitable for variable rate packet data. In order to support the mobile internet access, it is essential do design a reasonable packet switching network which supports the mobility. Since mobile packet network has longer latency, high speed switching and QoS are required to meet the user's requirements. In this paper, we suggest an resonable way to construct a network and its operation procedures utilizing GPRS(General Packet Radio Service) network and MPLS(Multi Protocol Label Switching) to provide a high speed switching and QoS mobile internet access. GPRS is used as a network which supports the mobility and MPLS guarantees the QoS and high speed IP protocol transmission based on the ATM switching technology.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.11
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• pp.2405-2410
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• 2012

The 4th generation mobile communication system, LTE, does not support an additional core network to provide voice service, and it is merged into a packet network based on all IP. Although Voice service over LTE can be supported by VoIP, it will be provided by the existing 3G networks because of the discontinuity of LTE coverage. However, it is inevitable to adopt VoIP over LTE to provide high quality voice service. In this paper, we investigate the capacity of VoIP over LTE. Our results indicate that spectral efficiency can be significantly improved as channel bandwidth increases in terms of VoLTE capacity. In addition, we can achieve higher VoLTE capacity without decreasing control channel capacity.

• 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).

• 한국정보통신설비학회:학술대회논문집
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• 2009.08a
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• pp.99-103
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• 2009

Recently, as wired and wireless communication services have rapidly developed and multimodal mobile devices which have various characteristics have widely spread, the need for new convergence services increases. The growing population of VoIP technologies and the high communication expense yield that the market of IP based telephony such as WiFi phone and IP phone is substituted for one of the conventional PSTN telephony. With the help of this trend, the wireline network operators desire to find a market in mobile networks. Therefore, they focus on Fixed Mobile Convergence (FMC) service as one of the key factors to accomplish this goal. FMC services are able to provide the mobility of voice services between circuit switched and packet switched networks. IP Multimedia Subsystem (IMS) based Voice Call Continuity (VCC) is one of the schemes to embody FMC services. As Application Server (AS) which has this VCC function provides seamless handover of services between heterogeneous networks, FMC subscribers can communicate seamlessly with others m WiFi domain and COMA domain using WiFi-COMA dual phone. Most of enterprises have already introduced IP network infrastructure and IP-PBX (Private Branch eXchange) for telephony. However, the problems of high communication cost and work inefficiency due to frequent outside jobs or business trips have remained. In order to solve these problems, demands for enterprise FMC services increase. In this paper, we introduce a new IP-PBX based VCC model that can provide seamless handover of voice services between WiFi and COMA networks for enterprise users and we investigate some interworking and security issues between Soft Switch (SSW) and IMS, or between IMSs. In addition, we introduce a new service that can provide the continuity of voice sessions as well as video sessions using Multimedia Session Continuity (MMSC) technology which has evolved from VCC. This service is expected to be one of the next-generation personalized services based on user's context.

• Journal of Broadcast Engineering
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• v.12 no.2
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• pp.80-92
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• 2007

IPTV Services that is representative union service of broadcasting and telecommunication need guarantee of QoS, efficiency of multicasting, and hish bandwidth on the network. Because typical TDM based metro transport network was designed by transporting fixed voice traffic with stable and recovering method, it has a defect of bottleneck and a waste of bandwidth for acceptance of data traffic with burst feature and then all of data are treated equally at the transport network because it cannot classify between advanced high end service and best effort low end service. for completely resolving this kind of problem about increasing burst traffic and QoS issues, firstly we need to new design for transport network. This paper presents transformation method from TDM based metro transport network to packet based transport network and advantage and effectiveness of packet based transport network and also indicates technical factor and characters about method of packet transport system. As a result of research, the Packet Transport System, which is a transmission network for packet delivery, take in not only a specific character of legacy TDM but QoS, Multicast and high bandwidth, then, it is able to keep an effective bandwidth and a stabilized performance of packet transmissions. Additionally, if a fault be occurred on an optical link, the system is able to guarantee a differential QoS by an each service class using an algorithm to make certain of a traffic existence and contain a protective mechanism.

• Journal of the Korea Society of Computer and Information
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• v.20 no.6
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• pp.29-35
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• 2015

The heterogeneous LTE (Long Term Evolution)-Advanced networks comprising a macrocell and femtocells can provide an efficient solution not only to extend macrocell coverage but also to deal with packet traffics increasing explosively within macrocells. An efficient resource management scheme is necessary to maintain the QoS (Quality of Service) of mobile multimedia services because the LTE-Advanced system should support not only voice but also mobile applications such as data, image and video. This paper proposes a resource management scheme to guarantee QoS continuity of multimedia services and to maximize the resource utilization in OFDMA (Orthogonal Frequency Division Multiple Access) based LTE-Advanced systems. This scheme divides the resources into several service blocks and allocates those resources based on the competition between macrocell and femtocell. Simulation results show that it provides better performances than the conventional one in respect of handover failure rate and blocking rate.