• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.7
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• pp.1744-1754
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• 1998

In this paper, the data service protocol which could support data service more efficiently between mobile host and fixed host in wire network is investigated. Wireless link has high bit error rate compared with wire link. Therefore, TCP performance for the data service is degraded in wire and wireless interworking environment. Thus, to reduce performance degradation. Interworking module withsimple protocol processing function is proposed. This, interworking module analyzes the hearder information of TCP fram. If received TCP freame is a duplicated frame, TCP frame is discared. Also, if interworking moudule receives retransmission request frame is a duplicated frame, TCP freme is discarded. Also, if interworking module receives retansmission request frame, interworking module performs retransmission procedure. According to simulation results, the proposed IWF shows better performance than traditional IWF in view of delay and throughput in the wire and wireless interworking environments.

• Journal of information and communication convergence engineering
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• v.9 no.4
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• pp.411-419
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• 2011

In this paper, we propose an Adaptive Regulating Drop (ARD) marker, as a novel dropping strategy at the ingressive edge router, to improve TCP fairness in assured services (ASs) without a decrease in the link utilization. To drop packets pertinently, the ARD marker adaptively changes a Temporary Permitted Rate (TPR) for aggregate TCP flows. The TPR is set larger than the current input IN packet rate of aggregate TCP flows while inversely proportional to the measured input OUT packet rate. To reduce the excessive use of greedy TCP flows by notifying droppings of their IN packets constantly to them without a decrease in the link utilization, the ARD marker performs random early fair remarking of their excessive IN packets to OUT packets at the aggregate flow level according to the TPR. In addition, an aggregate dropper is combined to drop some excessive IN packets fairly and constantly according to the TPR. Thus, the throughput of a TCP flow no more depends on only the sporadic and unfair OUT packet droppings at the RIO buffer in the core router. Then, the ARD marker regulates the packet transmission rate of each TCP flow to the contract rate by increasing TCP fairness, without a decrease in the link utilization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.7A
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• pp.700-709
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• 2007

The heterogeneous wireless access networks has been envisioned to characterize the future wireless networks. In such environments, TCP(Transmission Control Protocol) has to experience poor end-to-end performance because bandwidth and link delay change suddenly when a mobile node moves over different types of wireless networks, which is called vertical handoff. In this paper, we propose a new TCP which maintains each set of congestion control variables, which we call TCP context, for each type of wireless network. The proposed TCP can switch the TCP context against vertical handoff in order to adjust quickly to a newly arrived network. In simulations, the proposed TCP has higher throughput than TCP SACK(Selective Acknowledgment Options) due to its great features to vertical handoff situations.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.17-24
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• 2005

Traditional Internet applications such as FIP and E-mail are increasingly sharing bandwidth with newer, more demanding applications such as Web browsing, IP telephony, video conference and online games. These new applications require Quality of Service (QoS), in terms of delay, loss and throughput that are different from QoS requirements of traditional applications. Unfortunately, current Active Queue Management (AQM) approaches offer monolithic best-effort service to all Internet applications regardless of the current QoS requirements. This paper proposes and evaluates a new AQM technique, called MCDT that provides dynamic and separated buffer threshold for each Applications, those are FTP and e-mail on TCP traffic, streaming services on tagged UDP traffic, and the other services on untagged UDP traffic. Using a new QoS metric, our simulations demonstrate that MCDT yields higher QoS in terms of the delay variation and a packet loss than RED when there are heavy UDP traffics that include streaming applications and data applications. MCDT fits the current best-effort Internet environment without high complexity.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.2
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• pp.97-103
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• 2009

Improving TCP performance has long been the focus of many research efforts in 802.11 wireless networks study. Hop count and Round Trip Time (RTT) are the critical sources which serious affect the TCP performance on end to end connection. In this paper, we analytical derived the affection and based on the analysis we propose TCP should Change its Expected Value (TCP-CEV) when hop count and RTT change by setting a reasonable CWND change rate to improve the performance. The proposed scheme is applicable to a wide range of transport protocols using the basic TCP mechanism, and the protocol behavior is analytically tractable. We show that our simple strategy improves TCP performance at least over 12% in a chain topology, 4.9% in a grid topology and improve the TCP convergence.

• Journal of information and communication convergence engineering
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• v.2 no.1
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• pp.9-14
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• 2004

In this paper the modeling of the RLP layer in CDMA2000 is presented, which uses the NAK based ARQ scheme for Random Error Channels. The RLP performs a partial link recovery through limited number of RLP frame transmission in case of frame error. In case when the RLP fails due to excessive frame error, the control is passed on to the higher (TCP) layer. The TCP layer provides the complete end-to-end recovery. Thus the reliable performance at the TCP/RLP is essential to maintain the required Quality of Service in the DS-CDMA wireless links. The modeling is done for the performance analysis of the system in terms of the throughput and the mean extra delays, which are calculated analytically and are compared with the results generated by the simulations. This paper studies the effect of the random errors over different types of RLP frame formats and also the performance of the NAK based ARQ mechanism used under these conditions. The simulation provides with the over all characteristics of the throughput and the mean extra delay in terms of realistic environment parameters like Eb/No and probability of packet error (PE), based on the channel conditions.

• Journal of KIISE
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• v.41 no.11
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• pp.974-981
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• 2014

These days, use of multimedia streaming service and demand of QoS (Quality of Service) improvement have been increased because of development of network. QoS of streaming service is influenced by a jitter, delay, throughput, and loss rate. For guaranteeing these factors which are influencing QoS, the role of transport layer is very important. But existing TCP which is a typical transport layer protocol increases the size of congestion window slowly and decreases the size of a congestion window drastically. These TCP characteristic have a problem which cannot guarantee the QoS of UHD multimedia streaming service. In this paper, we propose a router buffer based congestion control method for improving the QoS of UHD streaming services. Our proposed scheme applies congestion window growth rate differentially according to a degree of congestion for preventing an excess of available bandwidth and maintaining a high bandwidth occupied. Also, our proposed scheme can control the size of congestion window according to a change of delay. After comparing with other congestion control protocols in the jitter, throughput, and loss rate, we found that our proposed scheme can offer a service which is suitable for the UDH streaming service.

• Journal of KIISE:Information Networking
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• v.29 no.4
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• pp.447-458
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• 2002

In this paper, we design and implement Wireless Transaction Protocol(WTP) and evaluate it for wireless transaction processing in mobile computing environments. The design and implementation of WTP are based on the coroutine model that might be suitable for light-weight portable devices. We test the compatibility between our product and the other products such as Nokia, Kannel and WinWAP For the evaluation of WTP, we use an Internet simulator that can arbitrary generate random wireless errors based on the Gilbert model. In our experiment, the performance of WTP is measured and compared to those of Transmission Control Protocol(TCP) and TCP for Transactions. The experiment shows that WTP outperforms the other two protocols for wireless transaction processing in terms of throughput and delay. Especially, WTP shows much higher performance In ease of high error rate and high probability of burst errors. This comes from the fact that WTP uses a small number of packets to process a transaction compared to the other two protocols and introduces a fixed time interval for retransmission instead of the exponential backoff algorithm. The experiment also shows that the WTP performance is optimized when the retransmission counter is set to 5 or 6 in case of high burst error rate.

• Journal of Korea Multimedia Society
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• v.12 no.11
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• pp.1521-1529
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• 2009

There is a serious unfairness problem between upstream and downstream flows of AP in IEEE 802.11 Wireless LAN. This problem is because Wireless LAN's DCF MAC protocol provides AP with equal channel access priority to mobile noded. Also, it makes this problem worse that the TCP's Data segment loss is more effective on throughput than the TCP's ACK segment. In this paper, we first make several simulations to analysis the unfairness in the various point of view and to find reasons of the unfairness. Also, this paper presents AP's scheduling scheme to alleviate the unfairness problem and evaluate the scheme through ns2 simulator.

• Journal of Communications and Networks
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• v.13 no.3
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• pp.277-285
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• 2011

Real-time multimedia streaming over the Internet is rapidly increasing with the popularity of user-created contents, Web 2.0 trends, and P2P (peer-to-peer) delivery support. While many homes today are broadband-enabled, the quality of experience (QoE) of a user is still limited due to frequent interruption of media playout. The vulnerability of TCP (transmission control protocol), the popular transport-layer protocol for streaming in practice, to the packet losses, retransmissions, and timeouts makes it hard to deliver a timely and persistent flow of packets for online multimedia contents. This paper presents TCP-real-time online multimedia environment (ROME), a novel transport-layer framework that allows the establishment and coordination of multiple many-to-one TCP connections. Between one client with multiple home addresses and multiple co-located or distributed servers, TCP-ROME increases the total throughput by aggregating the resources of multiple TCP connections. It also overcomes the bandwidth fluctuations of network bottlenecks by dynamically coordinating the streams of contents from multiple servers and by adapting the streaming rate of all connections to match the bandwidth requirement of the target video.