• Journal of Korea Multimedia Society
• /
• v.12 no.3
• /
• pp.409-418
• /
• 2009

Providing Quality-of-Service (QoS) guarantee requires for each router on the path of a traffic flow not to violate the flow's delay budget allocated to itself. Since the amount of load being offered to the router is determined by the budget, some imbalance in load among routers on the path may be alleviated by means of adjusting the budget. The equal allocation applied to the resource reservation protocol (RSVP) is simple to implement, but it has the drawback of a poor resource utilization. A load balancing method in which the delay budget being allocated to a router depends on its load state was developed to improve the drawback, but it's too complex to apply to the RSVP. This paper proposes an intra-path load balancing method not only applicable to the RSVP but also more effective in improving the drawback. The proposed method first partitions the end-to-end delay bound of a flow to routers by the RSVP and then let them adjust their budgets according to both the bottleneck state of the path and their links' bandwidth availabilities. The results of the simulation applying the proposed method to an evaluation network showed that the proposed method may provide the gain of 4 ${\sim}$ 17 % compared to that in the legacy one in terms of the number of maximally admittable flows.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.37 no.8B
• /
• pp.620-629
• /
• 2012

This paper proposes packet aggregation scheme for considering energy consumption of mobile devices on guaranteeing quality of voice. To our purpose, we analyze VoIP service on wireless multi-hop channel in terms of delay, packet loss and energy consumption when packet aggregation scheme is applied to VoIP service. Moreover, we induce a cost function with considering of tradeoff relation between quality of voice and energy consumption.

• The Journal of the Korea Contents Association
• /
• v.10 no.9
• /
• pp.36-44
• /
• 2010

In this paper, we propose video quality control system(VQCS) which can control video service quality through the monitoring of end-to-end available bandwidth for video streaming service like IPTV in NGN convergence network. Various multimedia services such as video, voice and gaming service can be provided by IPTV, and these services require large amounts of bandwidth. At this time, video quality degradation like video jerkiness, block distortion and blurring is caused when network available bandwidth is insufficient. Available bandwidth monitoring method is need to stably control video streaming quality. So, we periodically calculate the amount of the packets in link and measure available bandwidth by using total length field in IP header at terminal. Scalability extractor in network selects suitable video streaming data rate based on available bandwidth and transports video streaming with adaptive data rate to prevent video quality deterioration.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.11 no.6
• /
• pp.127-134
• /
• 2011

Recently, the wireless data communication infrastructure appearing before the footlights crosses the limit in the wire and develop into the fast speed. Particularly, the issue about the u-Healthcare service using this increases. And the investigation of the quality guarantee of the wireless data network is important for the smooth service providing. However, the quality control research about the wireless network is insufficient because the existing research performs the quality control between the termination about the wired network continually. This paper presented the mobile service quality measurement reference considering the radio environment,that is the method for satisfying the quality guarantee of the wireless data network user offered the u-Healthcare service as the research about this and user needs. For this purpose, wireless data quality measurement indicator and standard was selected. The quality measure evaluation plan was presented according to the service. And each service presented the quality index especially.

• Journal of the Korea Society of Computer and Information
• /
• v.10 no.4 s.36
• /
• pp.259-274
• /
• 2005

In this paper, we propose a methodology to estimate the number of users in e-Commerce systems. There have been a lot of previous work under the closed-LAN system environment. But the study on the number of acceptable users in real network environment is hard to find in the literature. Our research applies a Hybrid Simulation by using QoS results for end-to-end high-speed Internet service, and experiments are performed with regard to LAN and WAN, network equipments, and various network bandwidth. As result of the experiments we observed that the response time of high-speed Internet service media(Wireless LAN, ADSL, Cable, VDSL) depends heavily on the sequence and depth of transactions and on the ratio of transactional and non-transactional interactions. That is, as the network and application get more loads, the number of acceptable users decreases. By adding a cache server and an L4 switch to the simulation model, we analysed the changes in the number of users and client response time.

• The KIPS Transactions:PartC
• /
• v.18C no.6
• /
• pp.433-440
• /
• 2011

In this paper, we propose a new scheduling method that can guarantee reliable jitter by minimizing the queue length variation in the streaming service provisioning such as IPTV and VoD. The amount of traffic to be delivered within a certain time is very fluid because MPEG-4 and H.264 encoders use VBR(Variable Bit Rate) for delivering video streaming traffic. This VBR characteristic increases the end-to-end propagation delay variation when existing scheduling methods are used for delivering video frames. Therefore, we propose the new scheduling method that can minimize change rate of queue length by adaptively controling service rate taking into account the size of bulk incoming packets and arrival rate for bulk streaming traffic. Video frames can be more reliably transmitted through the minimization of the queue length variation using the proposed method. We use the queueing model and also carry out OPNET simulation to validate the proposed method.

• Proceedings of the Korea Information Processing Society Conference
• /
• 2009.04a
• /
• pp.1285-1287
• /
• 2009

VoIP 통신에서 단말간의 세션을 수립하고 종료하는 역할을 하는 SIP는 기본적으로 서버-클라이언트 방식으로 이루어져있다. 하지만 이는 확장성과 설치비용, 단일실패 지점 측면에서 큰 단점을 갖는다. 최근에 DHT를 이용한 구조적인 P2P를 SIP와 결합하여 위의 단점들을 극복하기 위한 연구들이 활발히 진행되고 있다. 하지만 DHT를 이용한 방식은 기존 서버-클라이언트 방식에 비해 검색지연이 늘어난다는 큰 단점을 가지고 있다. 검색지연은 SIP에서 QoS의 중요한 지표중 하나인 호 설정 지연에 60~80% 정도의 큰 영향을 미치게 되므로 해결해야할 중요한 문제이다. P2PSIP에서는 종단 노드가 검색한 사용자의 주소를 캐싱하여 같은 사용자를 재 검색시에 사용하고 있다. 이는 네트워크 전체 오버헤드 감소에 큰 영향을 주지 못할뿐만 아니라 발생된 검색비용을 자신만이 사용할 수 있으므로 비효율적이다. 이에 본 논문에서는 도메인으로 구분된 계층적 P2PSIP네트워크에서 프록시 서버의 역할을 하는 슈퍼노드를 이용한 효과적인 캐싱 방법을 이용하여 호 설정 지연을 단축할 수 있는 방안을 제안하다.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.30 no.10A
• /
• pp.886-895
• /
• 2005

As the networked media technology have been grown in recent, there have been many research works to deliver high-quality video such as HDV and HDTV over the Internet. To realize high-quality media service over the Internet, however, the network adaptive streaming scheme is required to adopt to the dynamic fluctuation of underlying networks. In this paper, we design and implement the network-adaptive HD(high definition) MPEG-2 streaming system employing the frame-based prioritized packetization. Delivered video is inputted from the JVC HDV camera to the streaming sewer in real-time. It has a bit-rate of 19.2 Mbps and is multiplexed to the MPEG-2 TS (MPEG-2 MP@HL). For the monitoring of network status, the packet loss rate and the average jitter are measured by using parsing of RTP packet header in the streaming client and they are sent to the streaming server periodically The network adaptation manager in the streaming server estimates the current network status from feedback packets and adaptively adjusts the sending rate by frame dropping. For this, we propose the real-time parsing and the frame-based prioritized packetization of the TS packet. The proposed system is implemented in software and evaluated over the LAN testbed. The experimental results show that the proposed system can enhance the end-to-end QoS of HD video streaming over the best-effort network.

• The KIPS Transactions:PartC
• /
• v.11C no.2
• /
• pp.235-244
• /
• 2004

TCP Vegas version provides better performance and more stable services than TCP Tahoe and Reno versions, which are widely used in the current Internet. However, in the situation where TCP Vegas and Reno share the bottleneck link, the performance of TCP Vegas is much smaller than that of TCP Reno. This unfairness is due to the difference of congestion control mechanisms of each TCP use. Several studies have been executed in order to solve this unfairness problem. In this paper, we analyze the minimum window size to maintain the maximum TCP performance of link bandwidth. In addition, we propose an algorithm which maintains the TCP performance and improves fairness by selective packet drops in order to allocate proper window size of each TCP connections. To evaluate the performance of the proposed algorithm, we have measured the number of data bytes transmitted between end-to-end systems by each TCP connections. The simulation results show that the proposed algorithm maintains the maximum TCP performance and improves the fairness.

• The Magazine of the IEIE
• /
• v.25 no.6
• /
• pp.69-78
• /
• 1998

프레임 릴레이망을 백본으로 한 LAN과 LAN접속으로 형성되는 기업통신망이 현재 활발히 전개되면서 인터넷을 통한 광고, 전자거래가 중요한 서비스로 진행되고 있다. 국내의 프레임 릴레이 서비스 현황을 살펴보면 KT, 데이콤, 일부기업등에서 프레임 릴레이 교환기를 수입하여 주고객인 기업에 서비스를 제공하고 있다. 이들은 사용자 요구에 의해 주로 LAN과 LAN에 접속된 라우터 등의 종단 시스템간 PVC(영구가상회선: Permanant Virtual Circuit)접속을 통하여 다양한 서비스를 제공하면서 독자적인 기업통신체계를 구축하고 있는 실정이다. 이와같이 사용자 요구로 PVC등록시 프레임 릴레이망과 LAN을 연결하는 링크의 QoS(서비스 질: Quality of Service), 트래픽 파라메터는 수동적으로 설정된다. 현재 국내에서는 프레임 릴레이 교환기 개발은 전무하고 단지 그들을 수입하여 운용하는 수준이며 향후 ATM(비동기 전달모드: Asynchronous Transfer Mode)망의 중요 서비스 연동이 될 프레임 릴레이 기술은 심도있게 연구되고 있다. 향후 고속통신을 제공하는 백본망은 기존의 프레임 릴레이망과 ATM망이 공존할 추세이며 점진적으로 프레임 릴레이망은 사라질 것이다. 이때 ATM 망이 기존의 프레임 릴레이망을 백본으로 한 기업통신망 수용시 원할한 서비스 제공을 위해서는 프레임 릴레이망을 백본으로 채택한 기존의 기업통신망을 ATM망에서 수용시 야기되는 문제점의 제시와 이의 해결은 매우 중요하다고 생각된다. FR/ATM(Frame Relay/ATM)연동시 기업통신망 입장에서는 PVC뿐만 아니라 SVC(교환가상회선: Switched Virtual Circuit)호처리가 수용되어야 한다. 이와같이 PVC와 SVC 수용으로 기업통신망은 통신망의 구성 및 관리가 복잡, 다양해지며, 또 그 기업은 관련된 다양하고 수많은 고객과 접속될 수 있어야 한다. 이때 ATM망을 백본으로 한 공중망이 관련 기업의 사설 정보를 관리하도록 한다는 것은 그 서비스의 이용자에게 보안문제를 제기함은 물론 서비스 이용자가 자신의 망에서 동작하는 여러 서비스를 고객 스스로가 관리하는 것이 바람직하다. CNN(고객망관리 : Customer Network Management)은 이러한 고객에 의한 통신망, 공중망에 대한 부분적 관리 및 서비스에 대한 관리가 이루어 질수 있도록 수행하는 기능이다. 본 논문에서는 ATM망이나 프레임 릴레이망과 ATM 망이 공존시 기업통신망관점에서 야기되는 문제점의 제시와 이의 해결 방안을 살펴보았다. 우선 FR/ATM 연동에서 PVC설정뿐만 아니라 SVC 설정 방법의 연구, 이때 고려된 QoS,트래픽 처리과정을 살펴보았다. 또한 망관리 입장에서 기업통신망 관리자와 ATM망 관리자와의 온-라인 서비스를 제공하기 위해서는 변화된 통신망 차원서의 확장된 Managed Object(MO)를 근간으로한 체계적인 CNN-NMS(CNM-Network management System)연동을 살펴본후 결론에서는 추후 계속연구 및 조사 되어야 할 항목을 기업통신망 입장에서 서술하여 보았다.