• Journal of Communications and Networks
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• v.7 no.3
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• pp.367-376
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• 2005

The smooth handoff supported by the route optimization extension to the mobile IP standard protocol should support a packet buffering mechanism at the base station (BS), in order to reduce the degradation in TCP performance caused by packet losses within mobile network environments. The purpose of packet buffering at the BS is to recover the packets dropped during intersubnetwork handoff by forwarding the packets buffered at the previous BS to the new BS. However, when the mobile host moves to a congested BS within a new foreign subnetwork, the buffered packets forwarded by the previous BS are likely to be dropped. This subsequently causes global synchronization to occur, resulting in the degradation of the wireless link in the congested BS, due to the increased congestion caused by the forwarded burst packets. Thus, in this paper, we propose an implicit priority forwarding (IPF) packet buffering scheme as a solution to this problem within mobile IP based networks. In the proposed IPF method, the previous BS implicitly marks the priority packets being used for inter-subnetwork handoff. Moreover, the proposed modified random early detection (M-RED) buffer at the new congested BS guarantees some degree of reliability to the priority packets. The simulation results show that the proposed IPF packet buffering scheme increases the wireless link utilization and, thus, it enhances the TCP throughput performance in the context of various intersubnetwork handoff cases.

• Journal of the Korea Society for Simulation
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• v.11 no.1
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• pp.1-10
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• 2002

IETF proposed Fast Handover mechanism tn reduce the latency during which the mobile node is effectively dusconnected from the Internet. In this paper, we propose three buffering strategies and a bicasting strategy to resolve packet loss and packet disordering problems in Fast Hangover mechanism. Through Java-based simulation we show thats, packet loss and disordering problems have been resolved by our proposal. As the result the buffering and bicasting strategies can reduce the handover delay caused by packet loss and packet disordering, in addition.

• Journal of the Institute of Electronics and Information Engineers
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• v.53 no.8
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• pp.19-27
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• 2016

Nowadays the adaptive buffering techniques for burst stream packet configuration and its operation algorithm to save energy in IP core network have been studied. This paper explains the selection method of packet buffering interval for energy saving when configuring burst stream packet at the ingress router in IP core network. Especially the adaptive buffering interval and its implementation scheme are required to improve the energy saving efficiency at the input part of the ingress router. In this paper, we propose the best adaptive buffering scheme that a current buffering interval is adaptively buffering scheme based on the input traffic of the past buffering interval, and analyze its characteristics of energy saving and end-to-end delay by computer simulation. We show the improvement of energy saving effect and reduction of mean delay variation when using an appropriate inverse-function selecting the buffering interval for the configuration of burst stream packet in this paper. We confirm this method have superior properties compared to other method. The proposed method shows that it is less sensitive to the various input traffic type of ingress router and a practical method.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.11 no.6
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• pp.89-94
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• 2012

In this paper, we considered the performances of PMIPv6 and DMM in the viewpoints of traffic load change and packet delivery latency time, where a new buffering in the previous MAAR is proposed to reduce packet loss during handover. To show the superiority of the DMM and to validate the operation of the buffering scheme, we accomplished its simulation under the typical handover. Our performance comparison results show that the DMM is better than the performance of traffic load change and packet delivery latency time of PMIPv6. We can see that the proposed buffering scheme is better than the existing one in terms of packet loss.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.7 s.337
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• pp.19-26
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• 2005

IETF proposed the Fast Handover mechanism to reduce the latency during which the mobile node is effectively disconnected from the Internet. However, the Fast Handover mechanism did not resolve packet loss and packet disordering problem. In this paper, we propose buffering mechanisms to resolve above problems in the Fast Handover mechanism. Though the simulation, we showed that packet loss and disordering problem have been absolutely resolved. In addition, our proposal reduced about 27$\%$ of the delay time by the buffering mechanism.

• Journal of Industrial Technology
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• v.20 no.B
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• pp.227-232
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• 2000

ISDN AO/DI provides efficient communication. It dynamically allocates channels for bandwidth, which is key component for packet data communication. In this paper, we analyze performance improvement of ISDN AO/DI system with buffering scheme and compare with AO/DI system without buffering scheme. From the simulation of AO/DI with buffering scheme, we see that the required bandwidth is less than of the AO/DI system without buffering scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.8B
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• pp.661-673
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• 2003

To prevent performance degradation of TCP due to packet losses in the smooth handoff by the route optimization extension of Mobile IP protocol, a buffering of packets at a base station is needed. A buffering of packets at a base station recovers those packets dropped during handoff by forwarding buffered packets at the old base station to the mobile user. But, when the mobile user moves to a congested base station in a new foreign subnetwork, those buffered packets forwarded by the old base station are dropped and TCP transmission performance of a mobile user in the congested base station degrades due to increased congestion by those forwarded burst packets. In this paper, considering the general case that a mobile user moves to a congested base station, we propose a Priority Packet Forwarding to improve TCP performance in mobile networks. In the proposed scheme, without modification to Mobile IP protocol, the old base station marks a buffered packet as a priority packet during handoff. And priority queue at the new congested base station schedules the priority packet firstly. Simulation results show that proposed Priority Packet Forwarding can improve TCP transmission performance more than Implicit Priority Packet Forwarding and RED (Random Early Detection) schemes.

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

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2013.10a
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• pp.619-621
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• 2013

This paper is proposing a real-time audio processing system for TCP/IP with server-client. The server sends the audio data packet which is the same size each time while playing the audio data. And the client plays the received audio data from the server. In general, The receiving speed of audio data packet is faster than processing the audio data. So, the unstable playback is occurred when playing the received audio data at the moment. In order to overcome this problem, the double buffering method is proposed.

• The Journal of the Korea institute of electronic communication sciences
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• v.16 no.6
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• pp.1161-1174
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• 2021

UDP is an unreliable data transmission protocol, in contrast to TCP, which implements reliable data delivery based on flow control and retransmission methods. However, the TCP operation algorithm for guaranteeing reliability is inefficient in improving transmission performance, and above all, there is no need to transmit data using the TCP method for fields that do not require perfect integrity. In this paper, we intend to discuss ways to improve the stability while maintaining the existing performance of UDP. To this end, a program applied with packet buffering and relaying techniques was developed, and the performance and stability of the experiment were verified.