• Journal of KIISE:Information Networking
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• v.31 no.6
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• pp.595-607
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• 2004

Networks often evolve to provide a host with multiple access points to the Internet. In this paper, we propose a transport layer load distribution mechanism utilizing the multiple network interfaces simultaneously. We specifically propose an extension of Stream Control Transmission Protoco1 (SCTP) to have load sharing over multiple network interfaces. We named the particular service provided by the Proposed load sharing mechanism to be LS (Load Sharing) mode service. LS mode service is based on the following four key elements: (i) the separation of flow control and congestion control, (ii) congestion window based striping, (iii) redundant packet retransmission for fast packet loss recovery, (iv) a novel mechanism to keep track of the receiver window size with the SACKS even if they arrive out-of-order. Through simulations, it is shown that the proposed LS mode service can aggregate the bandwidth of multiple paths almost ideally despite of the disparity in their bandwidth. When a path with a delay of 100% greater is utilized as the second path, the throughput is enhanced about 20%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.6B
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• pp.433-442
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• 2012

We have proposed a solution to multicast services and an advanced quality of service (QoS) mechanism on a packet transport system (PTS) based on PBB-TE. The point-to-multipoint (PtMP) connection in the PBB-TE system have been realized by grouping point-to-point (PtP) PTL trunks and mapping a BSI onto the PtP PTL trunks using a multicast backbone destination address. To provide end-to-end QoS of the PtMP services, the hierarchical QoS scheme for backbone service instances and connection-oriented paths has been implemented in the PTS. For providing different capabilities for service selection and priority selection, the PTS offers to customers three basic types of the port-based, C-tagged, and S-tagged service interface defined by the IEEE 802.1ah. To offer to customers different capabilities of the layer 3 applications and services, moreover, an IP-flow service interface have been added. In order to evaluate traffic performance for PtMP services in the PTS, the PtMP throughputs for the link capacity of 1 Gbps at the four service interfaces were measured in the leaves of the ingress edge node, the transit node, and the egress edge node. The throughputs were about 96 % because the B-MAC overhead of 22 bytes occupies 4% of the 512-byte packet. The QoS performance is ability to guarantee an application or a user a required bandwidth, and could be evaluated by the accuracy of policing or shaping. The accuracy of the policing scheme and the accuracy of the shaping scheme were 99% and 99.3% respectively.

• The Transactions of the Korea Information Processing Society
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• v.7 no.11S
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• pp.3723-3731
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• 2000

Recently, the growth of applications and services over high-speed Internet increase, ATM networks as wide area back-bone has been a major solution. The conventional TCP suite is still the standard protocol used to support upper application on current Internet and uses a window based protocol for flow control in the transport layer. When TCP data uses the UBR service in ATM layer, the control method is also buffer management. If a cell is discarded in ATM layer. one whole packet of TCP will be lost. Which is responsible for most TCP performance degradation and do not offer sufficiently QoS. To solve this problem, Several dropping strategies, such as Tail Drop, EPD, PPO, SPD, FBA, have been proposed to improve the TCP performance over ATM. In this paper, to improve the TCP fairness of end to end, we propose a packet dropping scheme algorithm using two fixed threshold. Under similar condition, we compared our proposed scheme with other dropping strategies. Although the number of VC is increased, simulation results showed that the proposed scheme can allocate more fairly each VC than other schemes.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.5
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• pp.2044-2062
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• 2018

Information theory progression along with the advancements being made in the field of Vehicular Ad hoc NETworks (VANETs) supports the use of coding-aware opportunistic routing for efficient data forwarding. In this work, we propose and investigate an adaptive coding-aware routing scheme in a specific VANET scenario known as a vehicular platoon. Availability of coding opportunities may vary with time and therefore, the accurate identification of available coding opportunities at a specific time is a quite challenging task in the highly dynamic scenario of VANETs. In the proposed approach, while estimating the topology of the network at any time instance, a forwarding vehicle contemplates the composition of multiple unicast data flows to encode the correct data packets that can be decoded successfully at destinations. The results obtained by using OMNeT++ simulator reveal that higher throughput can be achieved with minimum possible packet transmissions through the proposed adaptive coding-aware routing approach. In addition, the proposed adaptive scheme outperforms static transmissions of the encoded packets in terms of coding gain, transmission percentage, and encoded packet transmission. To the best of our knowledge, the use of coding-aware opportunistic routing has not been exploited extensively in available literature with reference to its implications in VANETs.

• Journal of KIISE
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• v.41 no.10
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• pp.810-823
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• 2014

The wireless mesh network based on IEEE 802.11s provides a routing based on a destination address as it inherits legacy internet architecture. However, this architecture interested in not 'what' which is originally the users goal but 'where'. Futhermore, because of the rapid increase of the number of mobile devices recently, the mobile traffic increases geometrically. It reduces the network effectiveness as increasing many packets which have same payload in the situation of many users access to the same contents. In this paper, we propose an OpenFlow-based contents routing for the wireless mesh network(WMN) to solve this problem. We implement contents layer to the legacy network layer which mesh network uses and the routing technique based on contents identifier for efficient contents routing. In addition we provide flexibility as we use OpenFlow. By using this, we implement caching technique to improve effectiveness of network as decreasing the packet which has same payload in WSN. We measure the network usage to compare the flooding technique, we measure the delay to compare environment using caching and non caching. As a result of delay measure it shows 20% of performance improve, and controller message decrease maximum 89%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.1B
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• pp.86-100
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• 2004

To provide the end-to-end service differentiation for assured services, the random early demotion and promotion (REDP) marker in the edge router at each domain boundary monitors the aggregate flow of the incoming in-profile packets and demotes in-profile packets or promotes the previously demoted in-profile packets at the aggregate flow level according to the negotiated interdomain service level agreement (SLA). The REDP marker achieves UDP fairness in demoting and promoting packets through random and early marking decisions on packets. But, TCP fairness of the REDP marker is not obvious as for UDP sources. In this paper, to improve TCP fairness of the REDP marker, we propose a modified REDP marker where we combine a dropper, meters and a token filling rate configuration component with the REDP marker. To make packet transmission rates of TCP flows more fair, at the aggregate flow level the combined dropper drops incoming excessive in-profile packets randomly with a constant probability when the token level in the leaky bucket stays in demotion region without incoming demoted in-profile packets. Considering the case where the token level cannot stay in demotion region without the prior demotion, we propose a token filling rate configuration method using traffic meters. By using the token filling rate configuration method, the modified REDP marker newly configures a token filling rate which is less than the negotiated rate determined by interdomain SLA and larger than the current input aggregate in-profile traffic rate. Then, with the newly configured token filling rate, the token level in the modified REDP marker can stay in demotion region pertinently fir the operation of the dropper to improve TCP fairness. We experiment with the modified REDP marker using ns2 simulator fur TCP sources at the general case where the token level cannot stay in demotion region without the prior demotion at the negotiated rate set as the bottleneck link bandwidth. The simulation results demonstrate that through the combined dropper with the newly configured token filling rate, the modified REDP marker also increases both aggregate in-profile throughput and link utilization in addition to TCP fairness improvement compared to the REDP marker.

• Journal of KIISE:Information Networking
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• v.32 no.4
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• pp.443-451
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• 2005

Stateful inspection devices must maintain flow information. These devices create the flow information also for network attack packets, and it can fatally inflate the dynamic memory allocation on stateful inspection devices under network attacks. The memory inflation leads to memory overflow and subsequent performance degradation. In this paper, we present a guideline to set the flow entry timeout for a stateful inspection device to remove harmful embryonic entries created by network attacks. Considering Transmission Control Protocol (TCP) if utilized by most of these attacks as well as legitimate traffic, we propose a parsimonious memory management guideline based on the design of the TCP and the analysis of real-life Internet traces. In particular, we demonstrate that for all practical purposes one should not reserve memory for an embryonic TCP connection with more than (R+T) seconds of inactivity where R=0, 3, 9 and $1\leqq{T}\leqq{2}$ depending on the load level.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.711-721
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• 2004

To provide the end-to-end service differentiation for assured services, the random early demotion and promotion (REDP) marker in the edge router at each domain boundary monitors the aggregate flow of the incoming in-profile packets and demotes in-profile packets or promotes the previously demoted in-profile packets at the aggregate flow level according to the negotiated interdomain service level agreement (SLA). The REDP marker achieves UDP fairness in demoting and promoting packets through random and early marking decisions on packets. But, TCP fairness of the REDP marker is not obvious as fur UDP sources. In this paper, to improve TCP fairness of the REDP marker, we combine a dropper with the REDP marker. To make packet transmission rates of TCP flows more fair, at the aggregate flow level the combined dropper drops incoming excessive in-profile packets randomly with a constant probability when the token level in the leaky bucket stays In demotion region without incoming demoted in-profile packets. It performs a dropping in the demotion at a domain boundary only if there is no prior demotion. The concatenate dropping at multiple domains is avoided to manifest the effect of a dropping at a domain boundary on TCP fairness. We experiment with the REDP marker with the combined dropper using ns2 simulator for TCP sources. The simulation results show that the REDP marker with the combined dropper improves TCP fairness in demoting and promoting packets by generating fair demoted in-profile traffic compared to the REDP marker. The effectiveness of the selected drop probability is also investigated with showing its impact on the performance of the REDP marker with the combined dropper.

• Journal of Korea Society of Industrial Information Systems
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• v.17 no.1
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• pp.31-38
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• 2012

Next generation cellular applications and smart phone usage generate very heavy wireless data traffic. It becomes ineluctable for mobile network operators to have multiple core network entities such as Serving Gateway and Packet Data Network Gateway in 4G-LTE to share this high traffic generated. A typical configuration consists of multiple serving gateways behind a load-balancer which would determine which serving gateway would service a end-users'request. Such hardware is expensive, has a rigid policy set, and is a single point of failure. Another perspective of today's increasingly high data traffic is that besides it is being widely accepted that the high bandwidth L TE provides is creating bottlenecks for service providers by the increasing user bandwidth demands without creating any corresponding revenue improvements, a hidden problem that is also passively advancing on the newly emerging 4G-LTE that may need more immediate attention is the network signaling traffic, also known as the control-plane traffic that is generated by the applications developed for smartphones and tablets. With this as starting point, in this paper, we propose a solution, by a new approach considering OpenFlow switch connected to a controller, which gains flexibility in policy, costs less, and has the potential to be more robust to failure with future generations of switches. This also solves the problem of scaling the control-plane traffic that is imperative to preserve revenue and ensure customer satisfaction. Thus, with the proposed architecture with OpenFlow, mobile network operators could manipulate the traffic generated by the control-plane signaling separated from the data-plane, besides also reducing the cost in installing multiple core-network entities.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.6B
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• pp.405-413
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• 2012

WFQ(Weighted Fair Queuing) is the most popular fair queuing algorithm to guarantee the Quality-of-Service(QoS), but it has the inherent drawback of a poor resource utilization, particularly under the low rate traffic requiring a tight delay bound. It was recently identified that the poor utilization is mainly due to non-optimized latency of a traffic flow and then LOFQ(Latency-Optimized Fair Queuing) to overcome the drawback was introduced. The LOFQ algorithm, however, renews their optimal latencies for all flows whenever a new flow arrives, which results in the high implementation complexity of O($N^2$).This paper is to reduce thecomplexity to O(1). The proposed method is first to derive the optimal latency index function from the statistical QoS characteristics of the offered load, and then to simply calculate the optimal latency index of the arriving flow using the function.