• Journal of the Korea Society for Simulation
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• v.10 no.2
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• pp.15-24
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• 2001

In this paper we study the delay patterns of voice traffic for end-to-end users Caused by serving the whole bunch of applications traffic at the same time on the Voice over Internet Protocol (VoIP) network. Given the current situation that voice traffic is served along with other application services on the VoIP network, it is quite necessary to figure out how and by what the voice traffic requiring high QoS is delayed. We compare the delay performance of voice traffic on the VoIP network under FIFO with the one under Weighted Fair Queuing(WFQ), and discover the differences of the delay performance resulting from the use of different voice codec algorithms. The results of our study show that using the voice codec algorithm with a higher coding rate nd the queuing algorithm of WEQ can provide users with high-quality voice traffic.

• Journal of Information Processing Systems
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• v.10 no.1
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• pp.119-131
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• 2014

In recent literature on traffic scheduling, the combination of the two-dimensional discrete-time Markov chain (DTMC) and the Markov modulated Poisson process (MMPP) is used to analyze the capacity of VoIP traffic in the cognitive radio system. The performance of the cognitive radio system solely depends on the accuracy of spectrum sensing techniques, the minimization of false alarms, and the scheduling of traffic channels. In this paper, we only emphasize the scheduling of traffic channels (i.e., traffic handling techniques for the primary user [PU] and the secondary user [SU]). We consider the following three different traffic models: the cross-layer analytical model, M/G/1(m) traffic, and the IEEE 802.16e/m scheduling approach to evaluate the performance of the VoIP services of the cognitive radio system from the context of blocking probability and throughput.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.5A
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• pp.441-450
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• 2006

Recently, with the exponential growth of the number of Internet users, IP traffic which occupies more than 90 percent of the entire Internet traffic affects significantly to the performance of networks. Therefore, the design of the self-similar traffic generator reflected the feature of IP traffic is very important to design the networks efficiently and evaluate the performance of it correctly. In this paper, we design the self-similar traffic generator using OPNET. In order to implement the self-similar characteristics, ON-OFF sources with Pateto distribution are employed and aggregated. The designed self-similarity traffic generator is evaluated and verified with R/S plot, variance time(VT) plot under the various offered loads and the number of sources. It is expected that the designed self-similar traffic generator can be put to practical use when wire or wireless networks is designed and verified as well as it can be useful to decide the specific parameter value for Internet traffic modeling.

• 한국정보통신설비학회:학술대회논문집
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• 2009.08a
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• pp.68-71
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• 2009

All-IP network requires change of the existing IP network engineering methods as the convergence service market between communication and broadcasting industries using IP network is growing rapidly. Especially the video services like IPTV require more strict transmission quality and higher bandwidth than the existing data services. So it is difficult to design All-IP network by the over-provisioning method which used to be used for the existing IP network design. It also requires a heavy investment which becomes one of big obstacles to the IPTV service expansion. In order to reduce the investment costs, it is required to design an optimized network by maximizing the utilization of the network resources and at the same time maintaining the customer satisfaction in terms of service quality. In this paper, we first analyze the effects of IPTV traffic on the existing internet. Then we compare two traffic engineering technologies, which are dimensioning without admission control and dimensioning with admission control, on the All-IP network design by simulation. Finally, we suggest cost effectiveness of traffic engineering technologies for designing the All-IP network.

• Journal of KIISE:Information Networking
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• v.29 no.1
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• pp.48-56
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• 2002

This paper proposes an IP(Internet Protocol) multicast routing method by multiple tree routes. Multiple trees, instead of a single tree, improve the quality of multicast services with nonlinear link cost and huge traffic demand. The proposed method adds tree routes until it satisfies target conditions, and it splits the multicast traffic demand into the chosen tree routes. We develop a mathematical model and optimal conditions for traffic splitting. The method works on the problems with many different simultaneous multicast traffic. Various experiments were carried and the results show that the new multicasting is fairly effective on end-to-end quality of services.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.1
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• pp.305-321
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• 2012

Since Internet Protocol (IP) is the most important protocol in Next Generation Networks (NGNs), 3rd Generation Partnership Project (3GPP) utilizes Session Initial Protocol (SIP) based on IP as the base protocol for negotiating sessions in IP Multimedia Subsystem (IMS). Different from traditional circuit-switched network, in IMS, the media traffic and signaling are delivered through IP transport. The media traffic may affect the signaling efficiency in core network, due to traffic collisions and best effort packets delivery. This paper proposes a novel path-migration mechanism for enhancing the traffic efficiency in integrated NGN-IMS. The simulation results show that the interference and traffic collision can be reduce by applying proposed path-migration mechanism and the signaling efficiency in core network can be improved with higher system capability and voice quality.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.5
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• pp.1109-1116
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• 2010

In this paper, some performance characteristics of VoIP(Voice over Internet Protocol) for MANET(Mobile Ad-hoc Networks) with simulation is studied and appropriate condition for implementation of VoIP service is suggested. VoIP simulator is implemented with NS(Network Simulator)-2. VoIP traffic for simulation is generated with some codecs of G.711, G.723.1, G.726-32, G.729A, GSM.AMR and iLBC. As simulation results for traffic transmission under $670{\times}670m$ 50node MANET environment, performance data for MOS(Mean Opinion Score), network delay, packet loss rate and transmission bandwidth are measured. Normalized analysis about measured results shows that maximum VoIP connection satisfying VoIP service quality condition is 15.

• 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.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.1
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• pp.81-89
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• 2005

In this paper, we have described the implementation of a practical simulation tool to design and analyze communication networks. Especially, this study is focused on the implementation and application methods of a simulator supporting VoIP The key characteristics of this particular system are its easy and intuitive usage, the real behaviors implementation of equipment and protocols, the actual generation and transmission of traffic for simulation, supporting of VoIP and so forth. Our system is distinguished from the existing tools which define only the nature of voice traffic, process those packets in the same way as general data, and analyze only the quality of packet transmission such as delay. Our tool presented in this paper generates and processes packets in different way according to the types of traffic distinguishing call signal from voice information traffic. Also, we equipped this system with the various devices such as VoIP gateway and gatekeeper, which enabled this system to analyze the performance of devices and the quality of voice traffic transmission between PSTN and Internet. By presenting the implementation methods and application of this system, we managed to propose the utilization scheme of a simulation tool.

• IEMEK Journal of Embedded Systems and Applications
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• v.5 no.1
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• pp.7-17
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• 2010

ITS(Intelligent Transport System) as things are used for Broadcast service using TDMB/TPEG/NAVI rather than personal seamless service. It is attaching weight to Traffic information gathering, Charging, Settlement service. This research is applied to improve DCCP(Datagram Congestion Control Protocol) which has function as protecting data and preserving message boundary. The improving method is like that we solve data trust in UDP because Connection and Transmission overhead in UDP is less than in TCP. We fix the data loss which is generated from unordered delivery section of IP base wireless service by using DCCP protocol. We guarantee of connection with OBE(On-Board Equipment) and reliance about transmission of data by complement to mapping table and multi-hoping. Finally, We evaluate the performance about transmission of IP based data. We constructed a test-bed near research center for this test.