• Journal of the Institute of Electronics Engineers of Korea TC
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• 제43권3호
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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 Korea Computer Industry Society
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• 제2권9호
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• pp.1163-1174
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• 2001

A general goal of the ATM(Asynchronous Transfer Mode) network is to support connections across various networks. ABR service using EPRCA(Enhanced Proportional Rate Control Algorithm) switch controls traffics in ATM network. EPRCA switch, traffic control method uses variation of the ACR(Allowed Cell Rate) to enhance the utilization of the link bandwidth. However, in ABR(Available Bit Rate) service, different treatments are offered according to different RTTs(Round Trip Times) of connections. To improve the above unfairness, this paper presents ABR DELAY mechanism, in which three reference parameters for cell delay are defined, and reflect on the messages of RM(Resource Management) cells. To evaluate our mechanism, we compare the fairness among TCP connections between ABR DELAY mechanism and ABR RRM mechanism. And also we execute simulations on a simple ATM network model where six TCP connections and a background traffic with different RTTs share the bandwidth of a bottleneck link. The simulation results, based on TCP goodput and efficiency, clearly show that ABR DELAY mechanism improves the fairness among TCP connections.

• Journal of KIISE
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• 제42권11호
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• pp.1322-1331
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• 2015

According to the hypervisor scheduler, the vCPU (virtual CPU) operates under two states: the running state and the stop state. When the vCPU is in the stop state, incoming events are delayed until that vCPU's state changes to the running state. The latency in handling such events that are sent to the vCPU is regarded as the I/O latency. Since a SMP (symmetric multiprocessing) VM (virtual machine) incorporates multiple vCPUs, the event latency on a SMP VM can vary according to specific vCPU that receives the event. In this paper, we propose a new scheme named event routing that sends events according to the operation state of each vCPU to reduce the event latency on an SMP VM. We implemented the proposed event routing scheme in Xen ARM hypervisor and confirmed the reduction of I/O latency from measuring the network RTT (round trip time) and the TCP bandwidth under a variety of testing conditions. The network RTT decreases by up to 94% and the TCP bandwidth increases up to 35% when compare to native Xen ARM.

• Journal of KIISE:Information Networking
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• 제32권3호
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• pp.359-369
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• 2005

Delay estimation is a difficult Problem in computer networks. Accurate one-way delay estimation is crucial because it serves a very important role in network performance and thus application design. RTT(Round Trip Time) is often used as an approximation of the delay, but because it is a sum of the forward and reverse delays, the actual one-way delay cannot be estimated accurately from RTT. To estimate one-way delay accurately, this paper proposes a new scheme that analytically derives one-way delay, forward and reverse delay respectively. We show that the performance of TCP can improve dramatically in asymmetric networks using our scheme. A key contribution of this paper is that our one-way deiay estimation is much more accurate than RTT estimation so that TCP can quickly find the network capacity in the slow start phase. Since RTT is the sum of the forward and reverse delays, our scheme can be applied to any protocol that is based on RTT.

• Journal of Digital Contents Society
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• 제16권3호
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• pp.381-388
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• 2015

TCP, which performs congestion control in congestion condition, is able to help a reliable transmission. However, packet loss can be increased because congestion window is increased by the time the packet is dropped in the process of congestion avoidance. In this paper, to solve the above problem, we propose a new congestion estimation based TCP congestion control scheme using the weighted average value of the RTT. After measuring a SRTT, which means the weighted average value of RTTs, at this point of time when a buffer overflow is occurred by an overloaded packet, the proposed scheme estimates the time, when the same SRTT is made in packet transmission, as a congestion time and then decreases the congestion window. The simulation results show that the proposed schem has a good performance in terms of packet loss rate and throughput when the packet loss due to buffer overflow is larger than that due to wireless channel.

• Journal of the Korea Society for Simulation
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• 제20권1호
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• pp.61-67
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• 2011

Smart TV System will lead to drastic change of communication and media industries as one of the emerging next generation network services. However, when the number of concurrent users increases rapidly, the issue of service quality degradation occurs because providing services to many users simultaneously stresses both the server and the network. The server limitation can be circumvented by deploying server clusters. but the network limitation is far less easy to cope with, due to the difficulty in determining the cause and location of congestion and in provisioning extra resources. In order to alleviate these problems, a number of schemes have been developed. Prior works mostly focus on reducing user-centric performance metrics of individual connection, such as the round-trip time(RTT), downloading time or packet loss rate, but tend to ignore the network loads caused by the concurrent connections or global network load balance. In this work, we make an in-depth investigation on the issue of user grouping for massive Smart TV services through simulations on actual Internet test-bed, PlanetLab.

• Journal of the Institute of Electronics Engineers of Korea TC
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• 제46권2호
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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.

• The KIPS Transactions:PartC
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• 제13C권3호
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• pp.353-358
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• 2006

Unlike TCP Reno, TCP Vegas recognizes network congestion through the measuring of RTT (Round Trip Time) and decides the main congestion control parameters, such as Windows size. But, congestion avoidance scheme of Vegas poorly reflects asymmetric characteristics of packet path because TCP Vegas uses the measuring of RTT that reflects forward/backward packet transmission delay as a forward delay. The RTT can't infer the forward/backward transmission delay variation because it only measures the packet's turn around time. In this paper, We have designed and implemented a new Vegas congestion control algorithm that can distinguish forward/backward network congestion. We have modified the source codes of TCP Vegas in Linux 2.6 kernel and verified their performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• 제30권4B호
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• pp.218-225
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• 2005

Although Snoop protocol can enhance TCP throughput efficiently in a wired-cum-wireless environment, it has a problem in performing local packet retransmissions under a burst error-prone wireless link. AT-Snoop protocol is proposed to cope with this Snoop protocol's problem by adopting adaptive timer. In this paper, TCP throughputs of AT-Snoop protocol have been analyzed with varying wireless link conditions and the ways of setting parameters of AT-Snoop protocol for higher TCP throughput are found out through computer simulations. From the simulation results, AT-Snoop protocol's two parameters, local retransmission threshold value and local retransmission timeout value, are closely related with the fading changing rate. To get higher TCP throughput, local retransmission threshold value and local retransmission timeout value should be set to a little bit larger values than average WSRTT(Wireless Smoothed Round Trip Time) and mean bad period of the wireless link, respectively.

• The Journal of Korean Institute of Communications and Information Sciences
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• 제34권5B호
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• pp.509-516
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• 2009

The Stream Control Transmission Protocol(SCTP) is a reliable transport layer protocol that provides several features. Multihoming is the one of the features and allows an association(SCTP's term for a connection) between two endpoints to use multiple paths. One of the paths, called a primary path, is used for initial data transmission and in the case of retransmission an alternate path is used. SCTP's current retransmission policy attempts to improve the chance of success by sending all retransmissions to an alternate destination address. However, SCTP's current retransmission policy has been shown to actually degrade performance in many circumstances. It is because that, due to Karn's algorithm, successful retransmissions on the alternate path cannot be used to update RTT(Round-Trip Time) estimation for the alternate path. In this paper we propose a scheme to avoid such performance degradation. We utilize 2bits which is not used in the flag field of DATA and SACK chunks to disambiguate original transmissions from retransmissions and to keep RTT and RTO(Retransmission Time-Out) values more accurate.