• Journal of the Korean Institute of Telematics and Electronics A
• /
• v.33A no.2
• /
• pp.1-7
• /
• 1996

In this paper, we analyzed the packet blocking probability of a finite buffer in a network interface unit. In general, a network interface unit which provides a means of interface between the network and computer has a microprocessor and a protocol processor for the network access protocols. It also has a receive buffer for the arriving packets from the network which is served by the microprocessor with service interruption by the protocol processor. In this paper, we modeled the receive buffer as a discrete time server with service interruption, and obtained the packet blocking probability using the mini-slot approximation.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.18 no.1
• /
• pp.110-122
• /
• 1993

When voice is transmitted through packet switching network, there needs a overload control, that is, a control for the congestion which lasts short periods and occurrs in local extents. In this thesis, we analyzed the performance of the statistical packet voice multiplexer using the overload control strategy by bit dropping. We assume that the voice is coded accordng to (4,2) embedded ADPCM and that the voice packet is generated and transmitted according to the procedures in the CCITT recomendation G. 764. For the performance analysis, we must model the superposed packet arrival process to the multiplexer as exactly as possible. It is well known that interarrival times of the packets are highly correlated and for this reason MMPP is more suited for the modelling in the viewpoint of accuracy. Hence the packet arrival process in modeled as MMPP and the matrix geometric method is used for the performance analysis. Performance analysis is similar to the MMPP IG II queueing system. But the overload control makes the service time distribution G dependent on system status or queue length in the multiplexer. Through the performance analysis we derived the probability generating function for the queue length and using this we derived the mean and standard deviation of the queue length and waiting time. The numerical results are verified through the simulation and the results show that the values embedded in the departure times and that in the arbitrary times are almost the same. Results also show bit dropping reduces the mean and the variation of the queue length and those of the waiting time.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.6 no.4
• /
• pp.1006-1025
• /
• 2012

Improving the quality of service in IP networks is a major challenge for real-time voice communications. In particular, packet arrival-delay variation, so-called "jitter," is one of the main factors that degrade the quality of voice in mobile devices with the voice-over Internet protocol (VoIP). To resolve this issue, a receiver-based enhanced timing recovery algorithm combined with active jitter estimation 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. Extensive simulations have shown that the proposed algorithm delivers high voice quality by pursuing an optimal trade-off between average buffering delay and packet loss rate.

• Journal of Korea Multimedia Society
• /
• v.20 no.9
• /
• pp.1527-1540
• /
• 2017

Recently, due to the rapid development of the internet, the production and demand of high capacity contents are rapidly increasing. In order to accommodate the changing trend of internet usage, researches on CCN, the future internet architecture, are actively being conducted. Mobility support is an important challenge in CCN real-time services, given that today's internet environment is changing to mobile and real-time services are increasing. In CCN real-time services, the mobility problem can be largely divided into consumer mobility, provider mobility, and intermediate-node mobility. Among them, when the intermediate-node moves in the CCN real-time services, the service disconnection occurs and the QoS degradation is caused. In this paper. we propose Intermediate-Node Mobility Support(INMS) to support the intermediate-node mobility in CCN real-time services. Experimental results show that the proposed scheme shows better performance than CCN in terms of service disconnection time and packet loss.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.22 no.10
• /
• pp.2191-2204
• /
• 1997

An efficient packet transmit scheduling algorithm should be able to allocate the resource to each connection fairly based on the bandwidth and quality of service requirements negotiated during the call admission procedure and it should be able to isolate the effects of users that are behaving badly. In this paper, we propose an effective round-robin packet transmit scheduling mechanism, which we call the delay tolerant packet reserving scheme (DTPRS) based on delay QoS requirments. The proposed scheme can not only provide fairness and but also reduce delay, delay variation, and packet loss rate by reserving output link time slots of delay tolerant packets and assigning the reserved slotsto delay urgent packets. Our scheme is applicable to high speed networks including ATM network because it only requires O(1) work to process a packet, and is simple enough to implement.

• Journal of Internet Computing and Services
• /
• v.6 no.3
• /
• pp.97-106
• /
• 2005

Generally, network QoS is determined by timeliness, bandwidth, and reliability, As IT technologies are growing day by day, the bandwidth and reliability factors are expected to be able to satisfy user's requirements. The timeliness factor, however, has not fully met user's requirements yet. In this paper, we propose a packet forwarding scheme named POSIA (Packet forwarding Scheme based on Interworking Architecture) that can provide quite a good QoS (quality of service) by minimizing processing time which is the major part of the timeliness factor. Based on path and resource reservation mechanism, the proposed POSIA makes routers on the packet forwarding path synchronize with each other and then forward packets, The POSIA has been verified to outperform the existing MPLS packet forwarding scheme through computer simulations using OPNET.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2009.10a
• /
• pp.966-968
• /
• 2009

In IEEE 802.16 system, it is important to allocate resources appropriately in order to the efficient utilization of resources among a variety of real-time and non real-time packets. although a lot of the real-time and non real-time packet scheduling schemes in OFDMA system have been proposed, it need to be modified to apply to IEEE 802.16 system. Since IEEE 802.16 supports five kinds of service classes, it is important to take QoS constraints of these classes into consideration. In this paper, we propose a efficient scheduling scheme to support various services by considering the QoS constraints of each classes.

• The KIPS Transactions:PartC
• /
• v.15C no.5
• /
• pp.391-398
• /
• 2008

To provide IPTV service over mobile networks, the mechanism that reduce packet loss and interrupt of multimedia service during the handover should be supported. Especially, buffering based mechanism is preferable for supporting IPTV services in the way of preserving streaming service using stored data and recovering non-received data after handover. But previous research doesn't consider the buffer allocation for applying various environments which can change handover time or end to end delay of relay node. This paper propose DBAHAJ mechanism that optimize buffer size of mobile nodes and relay node for supporting seamless IPTV service over mobile environments. Mobile node determines buffer size by checking handover time and maximum difference of sequence to keep playing video data. And multicast agent recovers packet loss during the handover by sending buffered data. By these two procedure, node supports seamless IPTV service on mobile networks. We confirm performance of this mechanism on NS-2 simulator.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.22 no.10
• /
• pp.2292-2302
• /
• 1997

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. In PRMA protocol, data terminals with random data packets cannot reserve slot. That is, slot reservation is applicable to the time constraint voice packet exclusively. But the reservation scheme has to be performed for loss sensitive data packet, and so data packets can get their quality of service. Therefore, in wireless MAC, reservation technique has to be used for both voice and data services. But in service aspects, if a fixed bandwidth is allocated to data terminals, time constraint voice packets may have a low efficiency. So in this study, the terminal which wants to request for ABR tyupe service, acquires a minimum bandwidth from system for thefirst time. If the system has extra available bandwidth, ABR terminals would acquire additional bandwidth slot by slot. As a result, APRMA protocol cansupporty the data service with loss sensitivity and maintain their channel utilization high. Also high priority services like voice can be satisfied with their QoS by APRMA.

• Proceedings of the Korea Society for Industrial Systems Conference
• /
• 2000.11a
• /
• pp.463-470
• /
• 2000

In this paper, we describe a wireless MAC protocol named APRMA(Abitrary Period Reservation Multiple Access), which is capable of supporting the ABR type data service and maximizing channel utilization. In original PRMA protocol, data terminals with random data packets cannot reserve slot. 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, so data packets can get their quality of service. The aspects of service, if fixed bandwidth is allocated to data terminals, time constraint voice packets may have a low efficiency So in this study, the terminal which wants to request for ABR type service, acquires a minimum bandwidth from system for the first time. If the system have 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 highly. Also high Priority services like voice can be satisfied with their QoS by APRMA.