• Journal of information and communication convergence engineering
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• v.9 no.5
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• pp.530-538
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• 2011

Performance of TCP can be severely degraded in Mobile IP-based wireless networks where packet losses not related to network congestion occur frequently during inter-subnetwork handoffs by user mobility. To solve such a problem in the networks using Mobile IP, the packet buffering method at a base station(BS) recovers those packets dropped during handoff by forwarding the buffered packets at the old BS to the mobile users. But, when the mobile user moves to a congested BS in a new foreign subnetwork, those buffered packets forwarded by the old BS are dropped and TCP transmission performance of a mobile user degrades severely. In this paper, we propose a Modified BLUE(MBLUE) buffer required at a BS to increase TCP throughput in Mobile IP-based networks. When a queue length exceed a threshold and congestion grows, MBLUE increases its packet drop probability. But, when a TCP connection is added at new BS by a handoff, the old BS marks the buffered packets. And new BS receives the marked packets without dropping. Simulation results show that MBLUE buffer reduces congestion during handoffs and increases TCP throughputs.

• ETRI Journal
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• v.33 no.1
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• pp.136-139
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• 2011

A modular upgrade design for packet transport switch nodes is presented where packet loss is dramatically reduced by intra-module and inter-module buffer sharing. This modular design offers significant cost and power reduction in a high-data-rate system where buffers are highly costly and power-greedy.

• Journal of Korea Multimedia Society
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• v.13 no.5
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• pp.740-753
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• 2010

Performance of TCP can be severely degraded in WLP-based Mobile IP wireless networks where packet loss not related to network congestion occurs frequently during WLP-based inter-subnetwork handoff by user mobility. To resolve such a problem in the networks using WLP-based Mobile IP, the packet buffering method recovering seamlessly the packets dropped due to user mobility has been proposed. The packet buffering method at a bridge station recovers those packets dropped during handoff by forwarding buffered packets at the old bridge station to the WLP device. But, when the WLP device moves to a congested bridge station in a new WLP foreign subnetwork, those buffered packets forwarded by the old bridge station are dropped and TCP transmission performance of a WLP device in the congested bridge station degrades due to increased congestion by those forwarded burst packets. In this paper, a PBM(Packet Bridge Marker) is proposed for preventing buffered out-of-profile(OUT) packets from reducing the throughput of in-profile(IN) packets of an Assured Service WLP device. From this operation, the packet losses of buffered OUT packets are avoided and the throughput of IN and Total packets of an AS WLP device are increased.

• Journal of KIISE:Computing Practices and Letters
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• v.15 no.9
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• pp.626-636
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• 2009

In Mobile IP-enabled wireless LAN (WLAN), packet flows are corrupted due to the handoff of a mobile node (MN) at the link and network layers, which results in burst packet losses and can cause temporary buffer underflow in a streaming client at the MN. This transient behavior hurts time-sensitive streaming media applications severely. Among many suggestions to address this handoff problem, few studies are concerned with empirical issues regarding the practical validation of handoff options on the time-sensitive streaming media applications. In this paper, targeting seamless streaming over Mobile IP-enabled WLAN, we introduce a seamless media streaming framework that estimates accurate pre-buffering level to compensate the handoff latency. In addition, we propose a link-layer (L2) assisted seamless media streaming system as a preliminary version of this framework. The proposed system is designed to reduce the handoff latency and to overcome the playback disruption from an implementation viewpoint. A packet buffering and forwarding mechanism with L2 trigger is implemented to reduce the handoff latency and to eliminate burst packet losses generated during the handoff. A pre-buffering adjustment is also performed to compensate the handoff latency. The experimental results show that the proposed approach eliminates packet losses during the handoff and thus verify the feasibility of seamless media streaming over Mobile IP-enabled WLAN.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5B
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• pp.500-511
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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 the wireless link utilization performance degrades due to increased congestion by those forwarded packets. In this paper, considering the case that a mobile user moves to a congested base station in a new foreign subnetwork, we propose an Implicit Priority Packet Forwarding to improve TCP performance in mobile networks. In the proposed scheme, the old base station marks a buffered packet as a priority packet during handoff. In addition, RED (Random Early Detection) at the new congested base station does not include priority packets in queue size and does not drop those packets randomly based on average queue size. Simulation results show that wireless link utilization performance of mobile hosts can be improved without modification to Mobile IP protocol by applying proposed Implicit Priority Packet Forwarding.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.6A
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• pp.832-844
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• 1999

In this paper, we analyzed mobile-IP protocol of IETF, and propose new method to improve performance in handover environments. In proposed method, cache agent manages mobility binding information for mobile host. Thus, effective support of mobility is possible. Also, when handover occurs, mobile host recognizes change of foreign agent, and transmits registration message to new foreign agent. However, during registration time, data packet loss is occurs in old foreign agent. Thus, we prevent data loss by using data packet buffering and forwarding in old foreign agent.According to simulation results for data packet transmission performance in the case of handover occurring, proposed method has better performance than previous method in the view of transmission delay and throughput. Especially, if handover occurs very often, data buffering and forwarding method in foreign agent could guarantee better performance.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.722-737
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• 2011

In this paper, we propose an estimation of network jitter that occurs when transmitting TCP packets containing MPEG-2 TS in progressive streaming service over wired or wireless Internet networks. Based on the estimated network jitter size, we can calculate required de-jitter buffering time to absorb the network jitter at the receiver side. For this purpose, by exploiting the PCR timestamp existing in the TS packet header, we create a new timestamp information that is marked in the optional field of TCP packet header to estimate the network jitter. By using the proposed de-jitter buffering scheme, it is possible to employ the conventional T-STD buffer model without any modification in the progressive streaming service over IP networks. The proposed method can be applicable to the recently developed international standard, MPEG DASH (dynamic adaptive streaming over HTTP) technology.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.4C
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• pp.353-364
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• 2002

To prevent the performance degradation of TCP due to packet loss in the smooth handoff by the route optimization extension of Mobile If protocol, a buffering of packets at a base station is needed. A buffering of packets at a base station recovers the packets dropped during the handoff by forwarding the buffered packets at the old base station to the mobile user. But, when the mobile user moves to a new foreign network which is connected to a congested router, the buffered packets forwarded by the old base station are dropped and the link utilization performance degraded due to increased congestion by the forwarded packets. In this paper, when the mobile user moves to a new foreign network which is connected to a congested router, Ive propose a packet forwarding control scheme required far the old base station to improve TCP performance in mobile networks. The old base station forwards or discards the buffered packets during handoff by proposed packet forwarding control scheme based on congestion states of RED(Random Early Detection) at the congested router. Simulation results slow that link utilization performance can be improved by applying proposed packet forwarding control scheme.

• Proceedings of the IEEK Conference
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• 2002.07b
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• pp.1094-1097
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• 2002

Realtime applications like VoIP(Voice over IP) in All-IP networks need smooth handoffs in order to minimize or eliminate packet loss as a Mobile Host(MH) transitions between network links. In this paper, we design a new dynamic buffering(DB) mechanism for IPv6 by which an MH can request that the router on its current subnet buffers packets on its behalf while the MH completes registration procedures with the router of a new subnet. Performance results show that our proposed buffering scheme with a dynamic buffer space allocation is quite appropriate for mobile Internet, or the All-IP environment in terms of the datagram loss rate.

• Proceedings of the IEEK Conference
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• 2001.06a
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• pp.297-300
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• 2001

This paper proposes a sound multicast packet-switching method which can less affect QoS degradation. The method includes a switch fabric with extra switching paths dedicated for multicast packets. Presented also are both a buffering structure and a scheduling algorithm for the proposed method. Simulation analysis for the method shows that the switching delay of unicast packets is decreased even though arrival rate of multicast packets is increased.