• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.28 no.1B
• /
• pp.34-44
• /
• 2003

As the recent rapid development of internet technology and the wide spread of multimedia communications, massive increase of network traffic causes some problems such as the lack of network paths and the bad quality of service To resolve these problems, this paper presents a web-based traffic control system which supports QoS of realtime packet transmission for the multimedia communication The traffic control system presented in this paper applies the integrated service model and provides QoS of packet transmission by means of determining the packet transmission rate according to the policies of network manager and the optimal resource allocation considering the end-to-end traffic load It also provides QoS for the realtime packet transmission through the admission controller and the packet scheduler by the modified $WF^2Q^+$ algorithm support asynchronous and class-based queuing.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.34 no.6A
• /
• pp.458-467
• /
• 2009

Long Tenn Evolution (LTE) is the next generation mobile phone technology which has being standardized by the Third Generation Partnership Project (3GPP). In the existing mobile communication networks, voice traffic is delivered through circuit switched networks. In LTE, however, all kinds of traffic are transferred through IP based packet switched networks which has best-effort characteristic. Therefore, providing QoS in LTE system is difficult. In order to provide QoS in LTE, RRM is very important. Especially, in part of RRM, call admission control (CAC) performs an important function to reduce network congestion and guarantee a certain level of QoS for on-going calls. In this paper, we propose a CAC algorithm in order to provide QoS for various kinds of services in LTE system. The performance of the proposed algorithm is evaluated with various simulation environments. The results show that the proposed algorithm provides QoS through rejections of requested calls. Especially, the proposed CAC algorithm can be satisfied with packet delay requirement defined in LTE specification.

• Proceedings of the Korean Information Science Society Conference
• /
• 2001.10c
• /
• pp.109-111
• /
• 2001

본 논문은 RTP/RTCP 표준 프로토콜을 이용하는 멀티미디어 스트리밍 환경에서 콘텐츠 특성에 영향받지 않고 프로토콜 수준의 캐슁을 지원하는 캐쉬 서버에 대하여 지술한다. 또한 RTP 패킷 단위로 캐슁된 콘텐츠에 대한 스트리밍 기법과 캐쉬 제어 기법을 기술한다.

• Journal of the Korea Academia-Industrial cooperation Society
• /
• v.15 no.4
• /
• pp.2382-2392
• /
• 2014

In this paper, a new data frame structure for wavelength routing in optical ring access network(ORAN) is designed and the functions of each field in the frame are defined. In addition, the IP packet transmission to the network layer corresponding to the newly proposed structure of the protocol stack is newly presented. Tha data transmission protocol design techniques of ORAN was proposed. IP packet and broadcasting packet transmission within the sub-network, and the IP packet transmission to other sub-network was shown through the process illustration. In the process, the encapsulation and framing process of the wavelength information to routing has been described in detail. And each step takes place in a packet transfer process is demonstrated. A ring type optical access network protocol is not yet research field. ORAN data transfer protocol to send/receive module structure of a subscriber node and a control node of the two rings which analyzed the results are shown. The high-speed Internet solution is proposed that by using wavelength routing the packet transport protocol for ORAN is designed.

• Journal of Internet Computing and Services
• /
• v.3 no.6
• /
• pp.43-51
• /
• 2002

In this paper, we study the new packet dropping scheme using an active queue management algorithm. Active queue management mechanisms differ from the traditional drop tail mechanism in that in a drop tail queue packets are dropped when the buffer overflows, while in active queue management mechanisms, packets may be dropped early before congestion occurs, However, it still incurs high packet loss ratio when the buffer size is not large enough, By detecting congestion and notifying only a randomly selected fraction of connection, RED causes to the global synchronization and fairness problem. And also, it is the biggest problem that the network traffic characteristics need to be known in order to find the optimum average queue length, We propose a new efficient packet dropping method based on the active queue management for congestion control. The proposed scheme uses the per-flow rate and fair share rate estimates. To this end, we present the estimation algorithm to compute the flow arrival rate and the link fair rate, We shows the proposed method improves the network performance because the traffic generated can not cause rapid fluctuations in queue lengths which result in packet loss

• Journal of the Korea Society for Simulation
• /
• v.17 no.4
• /
• pp.191-198
• /
• 2008

6LoWPAN (IPv6 Low-power Wireless Personal Area Network) is IPv6 packets transmission technology at Sensor network over the IEEE 802.15.4 Standard MAC and Physical layer. Adaptation layer between IP layer and MAC layer performs fragmentation and reassembly of packet for transmit IPv6 packets. RFC4944, IETF 6LoWPAN WG standard document define packet fragmentation and reassembly. In this paper, we propose the 6PASim (6LoWPAN Packet Simulator) to perform IPv6 packet fragmentation and reassembly for performance evaluation. The 6PASim consist of two parts. One is Packet_Transmit_module that makes IEEE 802.15.4 frames the IPv6 packet from upper layer, and transmit its. and the another is Packet_Receive_module that reassembles transferred frames and completes original IPv6 packets. we can evaluate frame transmit rate and amount of control message through 6PASim. The result of simulation shows the SRM (Selective Retransmission Method) scheme provider better performance than IRM (Immediate Retransmission Method) scheme.

• Proceedings of the Korea Contents Association Conference
• /
• 2006.11a
• /
• pp.75-79
• /
• 2006

In a VoIP(Voice over IP) system which support the realtime speech service, speech quality is deteriorated by the delay, the jitter, the loss, and the reversed packet order. In this thesis, APBC for receiver site is proposed, which compensate the jitter by the adaptive playout algorithm and conceal the packet loss, and align the packet order. Also, a VoIP application system is implemented, and the performance of APBC is verified on the implemented system by measuring the processing speed and the speech quality. From the result, processing speed is 257$\mu$sec, which is fast enough to deal with packet being received in realtime. Also, the speech quality by MOS(Mean Opinion Score) is improved as 18 percent compared with algorithm of fixed playout delay.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2000.10a
• /
• pp.307-312
• /
• 2000

In this paper, in order to improve the performance of TCP short traffic services in wireless Internet environments, the Split-ACKs(SPACK) scheme is proposed. In wireless networks, unlike wired networks, packet losses will occur more often due to high bit error rates. Therefore, each packet loss over wireless links results in congestion control procedure of TCP being invoked at the source. This causes severs end-to-end performance degradation of TCP. In this paper, to alleviate the TCP performance, the SPACK method split acknowledgement packets in the base station is proposed. Using computer simulation the performance of TCP using SPACK is analyzed and shows better performance than traditional TCP protocol.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.5 no.5
• /
• pp.923-930
• /
• 2001

In this paper, in order to improve the performance of TCP short traffic services in wireless Internet environments, the Split-ACKs(SPACK) scheme is proposed. In wireless networks, unlike wired networks, packet losses will occur more often due to high bit error rates. Therefore, each packet loss over wireless links results in congestion control procedure of TCP being invoked at the source. This causes severe end-to-end performance degradation of TCP. In this paper, to alleviate the TCP performance, the SPACK method, split acknowledgement packets in the base station, is proposed. Using computer simulation, the performance of TCP using SPACK is analyzed and shows better performance than traditional TCP protocol.

• Annual Conference of KIPS
• /
• 2000.10b
• /
• pp.1609-1612
• /
• 2000

무선망에서 기존의 회선 서비스뿐만 아니라 패킷 서비스 및 영상 서비스 등을 수용하는 데 있어서 고려되어야 할 것들 중 하나는 무선 자원의 활용이다. 특히 패킷 서비스의 경우 가변적인 패킷의 길이와 양으로 효율적인 무선 자원관리를 어렵게 하는 요소를 지니고 있다. 즉, 무선 자원을 효율적으로 운용하기 위해서는 동적인 무선 자원할당 메커니즘이 중요한 역할을 한다. UMTS 표준 문서에서는 계층 2 전송 프로토콜인 RLC (Radio Link Control) 버퍼의 양만을 측정하여 무선 자원 할당이 제어가 되도록 권고하고 있다. 하지만 트래픽 양의 변화가 큰 VBR 트래픽의 경우, 이와 같은 방법은 갑자기 증가하거나 감소하는 트래픽 양에 적응하기 힘들고 이는 무선 자원의 효율성을 저하시키며, QoS 제공도 어렵다. 본 논문에서는 먼저 트래픽 특성을 알 수 있는 토큰 버켓 모델을 RLC에 적용하고, 동적 무선 자원 할당을 위해 RLC 버퍼의 양뿐만 아니라 예상되는 유입 트래픽 양을 측정 및 보고하는 메커니즘을 제안한다.