• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.8
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• pp.1780-1784
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• 2015

Traditional IP routing, see only the Destination Address When Forwarding Layer 3 routing and exchange information and Destination-Based Routing Lookup is required for all Hop. Thus, all routers Full Internet routing information, the route information of more than about 120,000 may require. Therefore, the router configuration, which can be dispersed in the environment, the traffic load is required in accordance with this congestion. In this study, a unique characteristic of the Internet in the environment of an existing network Best Effect for QoS guarantee and hardware high speed switching of large multimedia data transmitted using a Labeling for forwarding a packet environment configuration is required. Video Stream Broadcast Transport Labeling rather than in much of the higher performance of the multi-step policy to most of the Video Stream Packet deulim was fixed to Labeling Header Format proposes a method of applying an effective QoS policy to a more simplified policy.

• The Transactions of the Korea Information Processing Society
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• v.7 no.4
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• pp.1211-1216
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• 2000

Asynchronous transfer mode (ATM) can be efficiently used to transport packet data services. The switching system will support voice and packet data services simultaneously from end to end applications. To guarantee quality of service (QoS) of the offered services, source rateot send packet data is needed to control the network overload condition. Most existing control algorithms are shown to provide the threshold-based feedback control technique. However, real-time voice calls can be dynamically connected and released during data services in the network. If the feedback control information delays, quality of the serviced voice can be degraded due to a time delay between source and destination in the high speed link. An adaptive algorithm based on the optimal least mean square error technique is presented for the predictive feedback control technique. The algorithm attempts to predict a future buffer size from weight (slope) adaptation of unknown functions, which are used fro feedback control. Simulation results are presented, which show the effectiveness of the algorithm.

• Journal of Internet Computing and Services
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• v.3 no.3
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• pp.67-74
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• 2002

Asynchronous transfer mode (ATM) can be efficiently used to transport packet data services. The switching system will support voice and packet data services simultaneously from end to end applications. To guarantee quality of service (QoS) of the offered services, source rate to send packet data is needed to control the network overload condition. Most existing control algorithms are shown to provide the threshold-based feedback control technique. However, real-time voice calls can be dynamically connected and released during data services in the network. If the feedback control information delays, quality of the serviced voice can be degraded due to a time delay between source and destination in the high speed link, An adaptive algorithm based on the optimal least mean square error technique is presented for the predictive feedback control technique. The algorithm attempts to predict a future buffer size from weight (slope) adaptation of unknown functions, which are used for feedback control. Simulation results are presented, which show the effectiveness of the algorithm.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.722-737
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• 2011

In this paper, we propose an estimation of network jitter that occurs when transmitting TCP packets containing MPEG-2 TS in progressive streaming service over wired or wireless Internet networks. Based on the estimated network jitter size, we can calculate required de-jitter buffering time to absorb the network jitter at the receiver side. For this purpose, by exploiting the PCR timestamp existing in the TS packet header, we create a new timestamp information that is marked in the optional field of TCP packet header to estimate the network jitter. By using the proposed de-jitter buffering scheme, it is possible to employ the conventional T-STD buffer model without any modification in the progressive streaming service over IP networks. The proposed method can be applicable to the recently developed international standard, MPEG DASH (dynamic adaptive streaming over HTTP) technology.

• Journal of Korea Multimedia Society
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• v.6 no.6
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• pp.1034-1045
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• 2003

In this paper, we study a multicast transport technique for multimedia services of mobile hosts in wireless network environments. To reduce packet loss during hand-off, we propose a Pre-join scheme in overlapped area and a Buffering scheme in crossover routers. To support seamless service in real time multimedia application, these scheme use an optimal path routing which was provided in remote subscription scheme and a prediction scheme of host movements which was considered overlapped area. To evaluate the peformance of our scheme, we compare Bi-direction tunneling of mobile If, Remote subscription, and MoM by using NS-2. As a result, our scheme shows better performance in network overhead, packet loss and bandwidth's use than other schemes.

• Journal of KIISE:Information Networking
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• v.37 no.5
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• pp.317-326
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• 2010

Nowadays, transmission bandwidth for network traffic is increasing and the type is varied such as peer-to-peer (PZP), real-time video, and so on, because distributed computing environment is spread and various network-based applications are developed. However, as PZP traffic occupies much volume among Internet backbone traffics, transmission bandwidth and quality of service(QoS) of other network applications such as web, ftp, and real-time video cannot be guaranteed. In previous research, the port-based technique which checks well-known port number and the Deep Packet Inspection(DPI) technique which checks the payload of packets were suggested for solving the problem of the P2P traffics, however there were difficulties to apply those methods to detection of P2P traffics because P2P applications are not used well-known port number and payload of packets may be encrypted. A proposed algorithm for identifying P2P heavy traffics based on flow transport parameters and behavioral characteristics can solve the problem of the port-based technique and the DPI technique. The focus of this paper is to identify P2P heavy traffic flows rather than all P2P traffics. P2P traffics are consist of two steps i)searching the opposite peer which have some contents ii) downloading the contents from one or more peers. We define P2P flow patterns on these P2P applications' features and then implement the system to classify P2P heavy traffics.

• The Transactions of the Korea Information Processing Society
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• v.6 no.11S
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• pp.3260-3268
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• 1999

As the proliferation of multimedia traffic over High-speed Internet increases, ATM network will be vital to adopt as backbone network over various parts of Internet. In this paper, we investigate the performance of TCP/IP traffic flow over ABR and UBR of ATM service to study for the high throughput and good fairness by simulation technique. Although TCP is run in the transport layer, it is controlled by several methods, e.g, EPD, PPD, RED, EFCI, ER etc, in ATM layer when TCP uses the ABR/UBR service. Therefore, if one cell is discarded in ATM layer, a packet of TCp will be laost. And, also, along with the increasing of the number of VC among switches, the throughput and fairness will be degraded. In order to improve these degradations, we propose the effective parameter control operations of EFCI and ER on ABR service, and also suggest the buffer management methods on UBR service. Finally, through the simulation results, the improved throughput and fairness are shown.

• Proceedings of the KIEE Conference
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• 2005.10b
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• pp.192-194
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• 2005

Acceptance of the international standards for video compression, such as H.261, MPEG-1 and MPEG-2, along with the developments in video codec hardware, has created an explosion of application. Among these, the long time quest for long-distance digital video transmission causes an increasing interest in transporting compressed video over networks which are nontraditional for this purpose, including asynchronous transfer mode networks, the Internet, and cellular and wireless channels. Transmission of compression video over packet network is improved for error resilience. And layered video coding techniques improves error resilience. We present a efficient method of scalable video coding for low bandwidth.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.2B
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• pp.352-360
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• 2010

Supporting user perceptual QoE-guaranteed IP-based multimedia service such as IPTV and Mobile IPTV, we represent an efficient QoE-aware transport stream assessment schemes to apply realtime mobile IPTV's contents distorted by various network errors such as bandwidth, delay, jitter, and packet loss. This paper proposes in detail an efficient matching and QoE-aware measurement methods. The brightness of the digitized contents per each frames of transport streams is used and applied to reduced-reference method. The hybrid video quality metric is designed by QoE-indicators such as blur, block, edge busyness, and color error. We compare original with processed source to evaluate them in a high precision degree of accuracy.

• Journal of Communications and Networks
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• v.13 no.3
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• pp.277-285
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• 2011

Real-time multimedia streaming over the Internet is rapidly increasing with the popularity of user-created contents, Web 2.0 trends, and P2P (peer-to-peer) delivery support. While many homes today are broadband-enabled, the quality of experience (QoE) of a user is still limited due to frequent interruption of media playout. The vulnerability of TCP (transmission control protocol), the popular transport-layer protocol for streaming in practice, to the packet losses, retransmissions, and timeouts makes it hard to deliver a timely and persistent flow of packets for online multimedia contents. This paper presents TCP-real-time online multimedia environment (ROME), a novel transport-layer framework that allows the establishment and coordination of multiple many-to-one TCP connections. Between one client with multiple home addresses and multiple co-located or distributed servers, TCP-ROME increases the total throughput by aggregating the resources of multiple TCP connections. It also overcomes the bandwidth fluctuations of network bottlenecks by dynamically coordinating the streams of contents from multiple servers and by adapting the streaming rate of all connections to match the bandwidth requirement of the target video.