• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.2
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• pp.392-401
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• 2007

Wireless network has high BER characteristic because of path loss, fading, noise and interference. Many packet losses occur without any congestion in wireless network. Therefore, many wireless TCP algorithms have been proposed. SNOOP, one of wireless TCP algorithms, hides packet losses for Fixed Host and retransmits lost packets in wireless network. However, SNOOP has a weakness for bust errors in wireless network. This paper proposes the SACK-SNOOP to improve TCP performance based on SNOOP and Freeze-TCP that use ZWA messages in wireless network. This message makes FH stop sending packets to MH. BS could retransmit error packets to MH for this time. SACK-SNOOP use improved Selective ACK, thereby reducing the number of packet sequences according to error environment. This method reduces the processing time for generation, transmission, analysis of ACK. This time gain is enough to retransmit local burst errors in wireless link. Furthermore, SACK-SNOOP can manage the retransmitted error by extending delay time to FH. The simulation shows that our proposed protocol is more effective for packet losses in wireless networks.

• Journal of the Institute of Convergence Signal Processing
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• v.2 no.3
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• pp.9-15
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• 2001

Very low bit rate video coding uses the inter-picture video coding method for high compression. The inter-picture video coding is coded based on the information of the previous frames so any packet loss can lead to reduce the image quality on the transmission. In this paper, we proposed the multi-stream generation method for inter-media synchronization of very low bit rate video based on TCP for reliable transmission. The proposed approach performs a reliable transmission via a TCP based protocol. This method incorporates multi-streams in order to enhance the robustness of delivery and can withstand against network jitter. Moreover, the client bandwidths are fully utilized in a highly efficient way.

• Proceedings of the KIEE Conference
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• 2005.10b
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• pp.397-399
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• 2005

Handover is one of the most important factors that may degrade the performance of TCP connections and real-time applications in wireless data networks. We proposed a seamless handover with Motion Prediction in IEEE 802.16e-based broadband wireless access networks. By intergrating MAC and network layer handovers efficiently, this scheme minimizes the handover delay and eliminates packet losses during handover Simulations show that this scheme achieves loss-free packet delivery without packet duplication and increases TCP throughput significantly.

• Journal of Communications and Networks
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• v.6 no.4
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• pp.376-386
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• 2004

In ad hoc networks, loss-based congestion window progression by the traditional means of duplicate ACKs and timeouts causes high network buffer utilization due to large bursts of data, thereby degrading network bandwidth utilization. Moreover, network-oriented feedbacks to handle route disconnection events may impair packet forwarding capability by adding to MAC layer congestion and also dissipate considerable network resources at reluctant intermediate nodes. Here, we propose a new TCP scheme that does not require the participation of intermediate nodes. It is a purely end-to-end scheme using TCP timestamps to deduce link conditions. It also eliminates spurious reductions of the transmission window in cases of timeouts and fast retransmits. The scheme incorporates a receiver-oriented rate controller (rater), and a congestion window delimiter for the 802.11 MAC protocol. In addition, the transient nature of medium availability due to medium contention during the connection time is addressed by a freezing timer (freezer) at the receiver, which freezes the sender whenever heavy contention is perceived. Finally, the sender-end is modified to comply with the receiver-end enhancements, as an optional deployment. Simulation studies show that our modification of TCP for ad hoc networks offers outstanding performance in terms of goodput, as well as throughput.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.9
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• pp.3502-3509
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• 2010

Streaming service in the wireless mobile network environment has been a very challenging issue due to the dynamic uncertain nature of the channels. Overhead such as congestion, latency, and jitter lead to the problem of performance degradation of an adaptive multi-streaming service. This paper proposes a AMSS (Adaptive Multi-level Streaming Service) mechanism to reduce the performance degradation due to overhead such as variable network bandwidth, mobility and limited resources of the wireless mobile network. The proposed AMSS optimizes streaming services by: 1) use of fuzzy similarity metric, 2) minimization of packet loss due to buffer overflow and resource waste, and 3) minimization of packet loss due to congestion and delay. The simulation result shows that the proposed method has better performance in congestion control and packet loss ratio than the other existing methods of TCP-based method, UDP-based method and VBM-based method. The proposed method showed improvement of 10% in congestion control ratio and 8% in packet loss ratio compared with VBM-based method which is one of the best method.

• The Journal of the Korea Contents Association
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• v.11 no.2
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• pp.69-78
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• 2011

Recently, the bandwidth available to an end user has been dramatically increasing with the advancing of network technologies. This high-speed network naturally requires faster and/or stable data transmission techniques. The UDT(UDP based Data Transfer protocol) is a UDP based transport protocol, and shows more efficient throughput than TCP in the long RTT environment, with benefit of rate control for a SYN time. With a NAK event, however, it is difficult to expect an optimum performance due to the increase of fixed sendInterval and the flow control based on the previous RTT. This paper proposes a rate control method on following a NAK, by adjusting the sendInterval according to some degree of RTT period which calculated from a set of experimental results. In addition, it suggests an improved flow control method based on the TCP vegas, in order to predict the network congestion afterward. An experimental results show that the revised flow control method improves UDT's throughput about 20Mbps. With combining the rate control and flow control proposed, the UDT throughput can be improved up to 26Mbps in average.

• Journal of KIISE:Information Networking
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• v.29 no.3
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• pp.242-252
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• 2002

When a Mobile IP handoff occurs, the packets in flight can be lost because they are tunneled based on out-of-date location information. In this parer, we propose an enhanced handoff protocol that achieves no packet loss during Mobile IP handoff over a wireless LAN. Our handoff protocol predicts the next foreign agent that a mobile host is to visit by using: the information from the data link layer of wireless LAN. After that, when a Mobile IP handoff occult, the current foreign agent forwards the packets destined to a mobile host to the predicted foreign agent which buffers them. This eliminates packet loss and reduces the packet forwarding delay. Our handoff protocol is simulated using the Network Simulator-2 (ns-2) and shows the substantial performance enhancement of TCP with much less overhead up to 6.2 times compared to standard Mobile IP.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.6 no.4
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• pp.216-223
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• 2013

As the fast growth of using Internet, the requests of clients having different types and pressing loads on the server have been increased in World Wide Web. Thus the interesting issue is how to measure the real capacity of a Web Server. There have been much recent studies about measuring the capacity of web server. But the cause of Server response time delay is not just server itself but also network packet loss. To measure the practical capacity of web server, we generate scalable clients using Posix Thread, transport packets which were generated by scalable clients to the server using UDP and receive the packets which were the remain packet from network packet loss using TCP. In this paper, we propose a method to measure the practical capacity of a web server using the Scalable Clients based on Posix Thread and the transport on Application level.

• Proceedings of the KIEE Conference
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• 2006.07d
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• pp.1828-1829
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• 2006

In this paper, we propose sliding mode congestion controller for differentiated-services network. Two important issue in differentiated-services architecture are bandwidth guarantee and fair sharing of unsubscribed bandwidth among TCP flows with and without bandwidth reservation. We use tight upper and lower bounds for various settings of differentiated-services parameters using the loss-bounded model. The Sliding mode congestion controller scheme is designed using nonlinear control theory based on a nonlinear model of the network that is generated using fluid flow consideration. The methodology used is general and independent of technology, as for example TCP/IP or ATM. The sliding mode congestion controller methodology has been applied to an TCP network. We use NS-2 simulation to demonstrate that the proposed control methodology achieves the desired behavior of the network, and possesses important attributes. as e.g, stable and robust behavior, high utilization with bounded delay and loss, together with good steady-state and transient behavior.

• Journal of KIISE:Information Networking
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• v.32 no.1
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• pp.20-28
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• 2005

TCP(Transmission Control Protocol) is one of the most important Internet protocols, which is widely used in wireless networks as well as wired networks. However, when TCP is deployed for wireless networks, it takes severe performance degradation because TCP was designed for wired network. To overcome this drawback, a number of TCP variants have been proposed in the literature. However, most previous schemes did not consider TCP enhancement over heterogeneous networks. In heterogeneous networks, an mobile node (MN) may move from one access network to another(i.e., vertical handover). In the case of vertical handover, an MN experiences a TCP performance degradation caused by the packet loss and the sudden change of link characteristics between two different access networks. In this work, we investigate the TCP performance degradation occurred in vortical handover across heterogeneous networks. First, we have conducted the measurement study over GPRS-WLAN testbed. In the measurement study. we observed the TCP performance degradation in the case of handover from WLAN to GPRS. In order to study more different TCP behaviors during vertical handover, we performed comprehensive simulations using a network simulator 2(ns-2). Based on measurement and simulation results, we investigated how to improve TCP performance in vertical handover and we concluded that the existing mechanisms cannot be perfect solutions and new mechanisms are strongly required.