• The Journal of Korean Institute of Communications and Information Sciences
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• v.11 no.3
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• pp.179-196
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• 1986

In this paper, the peformance of a statistical packet voice/data multiplexer is studied. In ths study we assume that in the packet voice/data multiplexer two separate finite queues are used for voice and data traffics, and that voice traffic gets priority over data. For the performance analysis we divide the output link of the multiplexer into a sequence of time slots. The voice signal is modeled as an (M+1) - state Markov process, M being the packet generation period in slots. As for the data traffic, it is modeled by a simple Poisson process. In our discrete time domain analysis, the queueing behavior of voice traffic is little affected by the data traffic since voice signal has priority over data. Therefore, we first analyze the queueing behavior of voice traffic, and then using the result, we study the queueing behavior of data traffic. For the packet voice multiplexer, both inpur state and voice buffer occupancy are formulated by a two-dimensional Markov chain. For the integrated voice/data multiplexer we use a three-dimensional Markov chain that represents the input voice state and the buffer occupancies of voice and data. With these models, the numerical results for the performance have been obtained by the Gauss-Seidel iteration method. The analytical results have been verified by computer simylation. From the results we have found that there exist tradeoffs among the number of voice users, output link capacity, voic queue size and overflow probability for the voice traffic, and also exist tradeoffs among traffic load, data queue size and oveflow probability for the data traffic. Also, there exists a tradeoff between the performance of voice and data traffics for given inpur traffics and link capacity. In addition, it has been found that the average queueing delay of data traffic is longer than the maximum buffer size, when the gain of time assignment speech interpolation(TASI) is more than two and the number of voice users is small.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.2
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• pp.81-89
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• 2008

Wideband speech, characterized by a bandwidth of about 7 kHz (50-7000 Hz), provides a substantial quality improvement in terms of naturalness and intelligibility. Although higher data rates are required, it has extended its application to audio and video conferencing, high-quality multimedia communications in mobile links or packet-switched transmissions, and digital AM broadcasting. In this paper, we present a new bandwidth-scalable coder for wideband speech and audio signals. The proposed coder spits 8kHz signal bandwidth into two narrow bands, and different coding schemes are applied to each band. The lower-band signal is coded using the ITU-T G.729/G.729E coder, and the higher-band signal is compressed using a new algorithm based on the gammatone filter bank with an invertible auditory model. Due to the split-band architecture and completely independent coding schemes for each band, the output speech of the decoder can be selected to be a narrowband or wideband according to the channel condition. Subjective tests showed that, for wideband speech and audio signals, the proposed coder at 14.2/18 kbit/s produces superior quality to ITU-T 24 kbit/s G.722.1 with the shorter algorithmic delay.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.11B
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• pp.1023-1030
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• 2003

In this paper, we deal with an effective dynamic bandwidth allocation(DBA) scheme in order to support the qualify of services (QoS) in the customer access network which supports various applications with own service requirements. we discuss the DBA scheme for upstream traffic in the EPON which support both the delay sensitive traffic such as voice and real-time video and non-real time traffic such as data and BE. We propose the new DBA scheme which guarantee the delay performance of the real time traffic and utilize the upstream bandwidth effectively in the limited resource environment. We analize the delay performance of the proposed scheme by simulation.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.10B
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• pp.630-638
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• 2005

As real-time media applications based on IP multicast networks spread widely, the end-to-end QoS (quality of service) provisioning for these applications have become very important. To guarantee the end-to-end QoS of multi-party media applications, it is essential to monitor the time-varying status of both network metrics (i.e., delay, jitter and loss) and system metrics (i.e., CPU and memory utilization). In this paper, targeting the multicast-enabled AG (Access Grid) group collaboration tool based on multi-Party real-time media services, a hybrid monitoring scheme that can monitor the status of both multicast network and node system is investigated. It combines active monitoring and passive monitoring approaches to measure multicast network. The active monitoring measures network-layer metrics (i.e., network condition) with probe packets while the passive monitoring checks application-layer metrics (i.e., user traffic condition by analyzing RTCP packets). In addition, it measures node system metrics from system API. By comparing these hybrid results, we attempt to pinpoint the causes of performance degradation and explore corresponding reactions to improve the end-to-end performance. The experimental results show that the proposed hybrid monitoring can provide useful information to coordinate the performance improvement of multi-party real-time media applications.

• The Journal of the Korea institute of electronic communication sciences
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• v.9 no.6
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• pp.695-701
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• 2014

Blackhole attacks having a unauthorized change of routing data will cause critical effects for transmission performance. The transmission performance will be improved to the a certain level by using or having IDS(Intrusion Detection System)/IPS(Intrusion Prevention System) as countermeasures to blackhole attacks. In this papar, the effects of IDS to ene-to-end performance of packet aggregation transmission are analyzed on MANET(Mobile Ad-hoc Network) with IDS under blackhole attacks. MANET simulator based on NS-2 is used to analyze performance parameters as MOS, connection ratio, delay and packet loss rate as standard performance parameters, an another performance factor which is suggested in this paper. VoIP(Voice over Internet Protocol) traffics, one of voice services, is used for performance analysis. A suggestion for IDS implementation on MANET with packet aggregations under blackhole is shown as one of results.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.11
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• pp.1481-1489
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• 2016

A art from the typical data services based on 1:1 data communications, various new services based on 1:N communications have recently appeared. These services are becoming to require advanced 1:N communication schemes which can transfer the same data to many receivers efficiently and in high-performance. Especially, a Push-To-Talk (PTT) service, which is an important service in public safety communication system, requires a service server to disseminate the same voice media data to multiple receivers in a group in real-time and low latency. In this paper, we propose an efficient scheme to disseminate the same data to multiple receivers in low latency. In addition, we provide an analysis which gives a guide the performance of the 1:N communications in practical wired/wireless system environments in the perspective of the PTT service index.

• Journal of Broadcast Engineering
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• v.10 no.2
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• pp.182-189
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• 2005

In this paper, an asymmetric WDM-EPON transmission scheme is proposed to be used in a high speed access network system, which is required to implement the convergence of broadcast and communication. WDM is used for downstream transmission from OLT to access nodes, satisfying wide bandwidth requirement for broadcasting and various multimedia services. And an EPON scheme, which is cheaper than WDM, is applied to upstream transmission where less bandwidth is required. A transmission test in physical layer was performed successfully and the results are provided. If ONUs are to be used in a home gateway, its protocol should be appropriate to its traffic pattern. Voice is sensitive to a time delay while data is not. A new dynamic bandwidth assignment protocol for PON system, which can cope with various types of data in access network is proposed and its performance is analysed. A maximum cycle time is specified to achieve the QoS of signals sensitive to time delay. And a minimum window is specified to prevent the downstream control signals from uprising. It is shown by simulation that the proposed EPON protocol can provide a better performance than previous ones.

• Journal of Internet Computing and Services
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• v.4 no.5
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• pp.61-68
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• 2003

In this paper, we propose a wireless MAC protocol named APRMA, which is capable of supporting the ABR type data service and Maximizing channel utilization. Data terminals with random data packets are not provided slot reservation with PRMA protocol. That is, slot reservation is applicable to the time constraint voice packet exclusively. But the reservation scheme have to be performed for loss sensitive data packet, and contended their quality of service, Therefore, in wireless MAC, reservation technique has to be used for both voice and data services. So the terminal which wants to request for ABR type service, is allocated a minimum bandwidth from system for the first time, If the system have some extra available bandwidth, ABR terminals would acquire additional bandwidth slot by slot, As a result, APRMA protocol can support the data service with loss sensitivity and maintain their channel utilization high.

• The Journal of the Korea Contents Association
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• v.6 no.3
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• pp.1-11
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• 2006

With the improvement of information telecommunication technology, the various service in broadband integrated services digital networks have a wide range of delay, delay jitter and cell loss probability requirements according to traffic specification. Therefore, the design of appropriate control schemes that can satisfy the cell loss, delay requirements with various traffic specification for B-ISDN is an extremely important challenging problem. In this paper, we propose a priority control scheme using a window counter and a cell counter per each type of class. In the proposed priority control scheme, for satisfying required service quality, we performed the priority control scheme using the delay/loss factors obtained by comparing window counter with cell counter. The performance of proposed control scheme is estimated by computer simulation. In the results of simulation, we verified that the proposed method satisfied per class requirements as the results showed that cell loss probability has a order of video, data, voice and delay time has a order of video, voice and data.