• Proceedings of the Korean Operations and Management Science Society Conference
• /
• 2005.05a
• /
• pp.230-237
• /
• 2005

WCDMA방식에서 채택한 AMR(Adaptive Multirate) 음성 코덱은 4.75kbps에서 12.2kbps까지 8개의 가변 출력률을 가진다. 기지국제어시스템(Radio Network Controller)은 무선망 상황에 따라 AMR 출력 모드를 가변적으로 조정해 줌으로써 높은 사용자 QoS와 효율적인 시스템 성능을 얻을 수 있다. 본 연구에서는 순방향 WCDMA 채널에서 세 가지의 AMR 출력 모드 할당 방식을 제안하고, 음성 사용자가 경험하는 QoS 만족도를 시스템 성능 척도로하여 제안된 방식들을비교 할 수 있는 분석적 방법을 제시한다. 실험 예제를 통해서 시스템 부하에 따른 시스템 성능 척도의 변화를 도시함으로써 최적의 AMR 모드 할당 방식을 결정하는 기준을 제시한다.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.10 no.4
• /
• pp.1768-1789
• /
• 2016

In contemporary video streaming systems based on HTTP protocol, video players at the client side are responsible for adjusting video quality to network conditions and user expectations. However, when multiple video clips are streamed simultaneously, an intricate application logic implemented in the video players overlays the TCP mechanism which is responsible for a balanced access to a shared network link. As a result, some video players may not obtain a fair share of network throughput and may be vulnerable to an unstable video bit-rate. Therefore, we propose to simplify the algorithms implemented in the video players, which are responsible for the adjustment of video quality and constrain their functionality only to sending feedback to a server about a state of the player buffer. The main logic of the system is shifted to the server, which is now responsible for bit-rate selection and prioritisation of the video streams transmitted to multiple clients. To verify our proposition, we performed several experiments in a laboratory environment which show that when the server cooperates with the clients, the video players experience fewer quality switches and the system achieves better fairness when allocating network throughput among the video players. However, this comes at the cost of worse utilisation of network bandwidth.

• The KIPS Transactions:PartC
• /
• v.8C no.1
• /
• pp.51-59
• /
• 2001

In order to support various QoS, IETF has proposed the Differentiated Services Model which provides discrimination service according to t the user’s requirements and payment intention intention for each traffic characteristic. This model is an excellent mechanism, which is not too c complicated in terms of the management for service and network model. Also, it has scalability that satisfies the requirement of Differentiated Services. In this paper, We define the Differentiated Services Model using feedback control, propose its control procedure, and analyze its p performance. In conventional model, non-adaptive traffic, such as UDP traffic, is more occupied the network resource than adaptive traffic, such a as TCP traffic. On the other hand, the Differentiated Services Model using feedback control fairly utlizes the network resources and even p prevents congestion occurrence due to its ability of congestion expectation.

• Journal of Korea Multimedia Society
• /
• v.5 no.4
• /
• pp.441-449
• /
• 2002

For multimedia traffics to be supported successfully in wireless network environment, it is necessary to provide Qualify-of-Service(QoS) guarantees among mobile hosts(clients). In order to guarantee the QoS, we have to keep the call blocking probability below a target value during hand-off session. However, the QoS negotiated between the client and the network may not be guaranteed due to lack of available channels for traffic of a new cell, since on service mobile clients should be able to continue their sessions. In this paper, we propose an efficient load-balancing algorithm based on the adaptive bandwidth reservation scheme for enlarging available channels in a cell. Proposed algorithm predicts the direction of clients in a cell and adjusts the amount of the channel to be reserved according to the load status of the cell. This method is used to reserve a part of bandwidths of a cell for hand-off calls to its adjacent cells and this reserved bandwidth can be used for hand-off call prior to new connection requests. If the number of free channels is also under a low threshold value, our scheme use a load-balancing algorithm with an adaptive bandwidth reservation. In order to evaluate the performance of our algorithm, we measure metrics such as the blocking probability of new calls and dropping probability of hand-off calls, and compare with those of existing schemes.

• Journal of KIISE:Information Networking
• /
• v.37 no.5
• /
• pp.363-373
• /
• 2010

The existing QoS mechanisms for video streaming are short of the consideration for various user environments and the characteristic of streaming applying programs. In order to overwhelm this problem, studies on the video streaming protocols exploiting scalable video coding (SVC), which provide spatial, temporal, and qualitative scalability in video coding, are progressing actively. However, these protocols also have the problem to deepen network congestion situation, and to lower fairness between other traffics, as they are not equipped with congestion control mechanisms. SVC based streaming protocols also have the problem to overlook the property of videos encoded in SVC, as the protocols transmit the streaming simply by extracting the bitstream which has the maximum bit rate within available bandwidth of a network. To solve these problems, this study suggests TCP-friendly network adaptive SVC streaming(T-NASS) protocol which considers both network status and SVC bitstream property. T-NASS protocol extracts the optimal SVC bitstream by calculating TCP-friendly transmission rate, and by perceiving the network status on the basis of packet loss rate and explicit congestion notification(ECN). Through the performance estimation using an ns-2 network simulator, this study identified T-NASS protocol extracts the optimal bitstream as it uses TCP-friendly transmission property and perceives the network status, and also identified the video image quality transmitted through T-NASS protocol is improved.

• Proceedings of the Korea Contents Association Conference
• /
• 2006.11a
• /
• pp.716-720
• /
• 2006

In this paper, we designed and proposed new routing algorithm that can support a variety of vital-sign traffic characteristic and could be applicable to USN for telemedicine by using adaptive transmission power level and increase frequency of routing request message. In proposed routing algorithm, when an emergency vital-sign traffic is applied, we use large transmission power to reduce route query response time and make the priority order in route process. On the other hand, for non emergency vital-sign traffic, we use low transmission power and adaptive decrease frequency of routing request message. which is insensitive to delay. The proposed scheme should be better QoS performance in complex USN than conventional method, which is performed based on uniform transmission power level.

• Journal of Korea Multimedia Society
• /
• v.6 no.5
• /
• pp.831-841
• /
• 2003

In wireless ATM network, Mobile terminals must offer function and QoS such as terminal of wire ATM network. The fast notification of BS (Base Station) scheduler about ATM cells arriving in mobile terminals is a critical item of the ATM-based centrally controlled MAC protocol. But, Collision is arisen at process that several mobile terminals require data transmission by contention method through uplink and because of re-transmission by collision, throughput of decrease and transmission delay increases. In this paper, We propose the MAC protocol that decide optimized request mini-slot's number considering number of request terminal predicted as statistical and its posterior throughput. According to simulation result, could know that proposed protocol shows high performance delay and throughput.

• Journal of Internet Computing and Services
• /
• v.10 no.2
• /
• pp.161-169
• /
• 2009

The development of Wireless Multimedia Sensor Networks(WMSNs) is getting realized in accordance with the increased demands to provide multimedia data transmission service such as CCTV movie, image chapter information, audio information based on wireless sensor network. It is very important to provide the differentiated quality assurance of service. In this paper, we propose a sensor medium access control protocol based on DSMAC, which provides differentiated Quality of Service(QoS) for delay. A proposed protocol is able to reduce the delay without increasing the energy consumption by adaptively changing the duty cycle according to the buffer occupancy. Simulation results show that the new MAC protocol performs better in terms of latency than S-MAC and DSMAC.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.10 no.3
• /
• pp.1071-1090
• /
• 2016

Most of scheduling methods in the Grid only consider one special attribute of the resource or one aspect of QoS (Quality of Service) of the job. In this paper, we focus on the problem that how to consider two aspects simultaneously. Based on the requirements of the jobs and the attributes of the resources, jobs are categorized into three kinds: CPU-overload, memory-overload, and bandwidth-overload jobs. One job may belong to different kinds according to different attributes. We schedule the jobs in different categories in different orders, and then propose a scheduling method-MTS (multiple attributes scheduling method) to schedule Grid resources. Based on the comparisons between our method, Min-min, ASJS (Adaptive Scoring Job Scheduling), and MRS (Multi-dimensional Scheduling) show: (1) MTS reduces the execution time more than 15% to other methods, (2) MTS improves the number of the finished jobs before the deadlines of the jobs, and (3) MTS enhances the file size of transmitted files (input files and output files) and improves the number of the instructions of the finished jobs.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.38 no.1
• /
• pp.1-13
• /
• 2001

In this paper, we propose a new dynamic bandwidth allocation and call admission control for VBR video sources with QoS constraints to support an efficient resource management and at the same Lime to satisfy the user's quality or service requirements. For the dynamic bandwidth allocation, first the next amount of traffic is predicted using a modified adaptive linear prediction method that considers abrupt scene change effects. And then, we dynamically allocate the necessary bandwidth to each connection based on the currently provided quality degree by the network with respect to the user's QoS requirements in terms of average delay and loss ratio. For the admission control, we determine the acceptance or rejection or a new connection based on the quality satisfaction degrees of the existing connections. Simulation results show that our proposed dynamic schemes are able to provide a stable service, which well meets the user's quality requirements.