• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.3 s.345
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• pp.126-134
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• 2006

Traditional TCP implementations have the under-utilization problem in large bandwidth delay product networks especially during the startup phase. In this paper, we propose a delay-based congestion control(DCC) mechanism to solve the problem. DCC is subdivided into linear and exponential growth phases. When there is no queueing delay, the congestion window grows exponentially during the congestion avoidance period. Otherwise, it maintains linear increase of congestion window similar to the legacy TCP congestion avoidance algorithm. The exponential increase phase such as the slow-start period in the legacy TCP can cause serious performance degradation by packet losses in case the buffer size is insufficient for the bandwidth-delay product, even though there is sufficient bandwidth. Thus, the DCC uses the RTT(Round Trip Time) status and the estimated queue size to prevent packet losses due to excessive transmission during the exponential growth phase. The simulation results show that the DCC algorithm significantly improves the TCP startup time and the throughput performance of TCP in large bandwidth delay product networks.

• Journal of the Institute of Convergence Signal Processing
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• v.7 no.3
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• pp.136-142
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• 2006

Existing buffer management schemes for shared-memory output queueing switches can be classified into two types: In the first type, some constant amount of memory space is guaranteed to each virtual queue using static queue thresholds. The static threshold method (ST) belongs to this type. On the other hand, the second type of approach tries to maximize the buffer utilization in 머 locating buffer memories. The complete sharing (CS) method is classified into this type. In the case of CS, it is very hard to protect regular traffic from mis-behaving traffic flows while in the case of ST the thresholds can not be adjusted according to varying traffic conditions. In this paper, we propose a new buffer management method called weighted dynamic thresholds (WDT) which can process packet flows based on loss priorities for quality-of-service (QoS) functionalities with fairly high memory utilization factors. We verified the performance of the proposed scheme through computer simulations.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.4
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• pp.741-746
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• 2012

Transmission control protocol(TCP) widely used as a transport protocol in the Internet includes a loss recovery function that detects and recovers packet losses by retransmissions. The loss recovery function consists of the two algorithms; fast retransmit and fast recovery. There have been researches to avoid nonnecessary retransmission timeouts (RTOs), which leads to selective acknowledgement (SACK) option and limited transmit scheme that are standardized by IETF (Internet Engineering Task Force). Recently, a method that covers the case in which a retransmitted packet is lost again has been propsed. The method, however, is not proved in terms of the additive increase multiplicative decrease (AIMD) principle of TCP congestion control. In this paper, therefore, we analyzed the method in terms of the principle by ns-simulations.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.1
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• pp.9-19
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• 1999

An increase of the number of the real time applications and the Internet hosts make the Internet architecture changed. Current Internet architecture has some problems to process the real time traffics. To solve these problem, a new Internet architecture is proposed as the Integrated Service model. In the current Internet, as multicast protocols, the QoS multicast and the best-effort multicast have been studied in their separate network environments. But, the Integrated Service Packet Network is a heterogeneous network composed of the QoS delivery domain and the best-effort delivery domain. Thus, those separated multicast protocols have limitations in an ISPN. In this paper, we propose a multicast protocol for the ISPN with the QoS and the best-effort multicasting, and analyze the performance of this protocol. As a result, we find that the packet losses are same for hybrid multicast and best-effort multicast when the bandwidth is sufficient. But, if there exist some background traffics, the hybrid multicast has less packet loss than of the best-effort multicast.

• 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.

• Journal of KIISE:Information Networking
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• v.33 no.3
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• pp.269-276
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• 2006

This paper proposes a seamless handover scheme for high-speed trains using dual mobile routers to minimize service disruption time and packet loss during handovers. In the proposed scheme, each of the dual mobile routers is located at each end of the moving network for space diversity. One of the two mobile routers can continuously receive packets from its home agent, while the other is undergoing a handover, but they act as one logical mobile router. Analytical and simulation results showed that the proposed scheme could provide no service disruption or packet losses during handovers.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.3
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• pp.111-118
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• 2013

The connection based TCP protocol provides reliable packet delivery, but it takes certain overhead compare to the connection-less UDP protocol. Thus, 3GPP devise the upper-layer RLC protocol which provides reliability based upon the UDP protocol. Consequently, each implement of a base station and/or mobile terminal require the development of the RLC protocol, and it must qualify various interoperable tests. In this paper, we implement a testbed which verifies the RLC protocol under various packet losses/inversion circumstances of networks. Finally, we propose our testbed as the RLC protocol tester for the developments.

• The KIPS Transactions:PartC
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• v.17C no.6
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• pp.451-458
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• 2010

To detect and recover packet losses over wireless links is very important in terms of reliability in packet transmission. Most wireless communication systems adopt an automatic repeat request (ARQ) protocol operating at link layer. However, it has been constantly addressed that the interaction not harmonized sufficiently between ARQ and TCP rather degrades TCP performance. In this paper, therefore, we propose an improved scheme from the aspect of the interaction with TCP loss recovery mechanism that can be applied to selective repeat ARQ (SR-ARQ) protocol and prove that the proposed scheme improves TCP performance significantly by OPNET simulations.

• The Journal of the Korea Contents Association
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• v.5 no.3
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• pp.201-208
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• 2005

To guarantee the quality of service for user request, we should consider various kinds of things. The important thing of QoS is that response time of service is transparently suggested 'to network users. We can know the response time of service using the information of network latency, system latency, and software component latency, In this paper, we carried out the modeling of network latency and analyzed the effects of packets loss to the network latency, Also, we showed the effectiveness of modeling using the NS-2. This research can help to provide the effective methods in case of SLA(Service Level Agreement) agreement between service provider and user.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2000.10a
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• pp.307-312
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• 2000

In this paper, in order to improve the performance of TCP short traffic services in wireless Internet environments, the Split-ACKs(SPACK) scheme is proposed. In wireless networks, unlike wired networks, packet losses will occur more often due to high bit error rates. Therefore, each packet loss over wireless links results in congestion control procedure of TCP being invoked at the source. This causes severs end-to-end performance degradation of TCP. In this paper, to alleviate the TCP performance, the SPACK method split acknowledgement packets in the base station is proposed. Using computer simulation the performance of TCP using SPACK is analyzed and shows better performance than traditional TCP protocol.