• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.7
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• pp.1330-1338
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• 2015

In wireless communication network, the amount of packet data for the video streaming in the playout buffer of the receiver is changed with time according to network condition. If the amount of packet data is less than a specific buffer amount, the buffer underflow problem is generated. On the contrary, if the amount of packet data is more than a given buffer amount, the buffer overflow problem is occurred. When the playout of the video streaming is processed, these buffer underflow and overflow problems cause stop and skip phenomenons and then provide the discontinuity of playout. Therefore, to solve the buffer underflow and overflow problems of the video streaming in wireless communication network, This paper analyzes the combined effect of Token Bucket scheme, which controls the bursty traffic, and AMP(Adaptive Media Playout) scheme, which adaptively changes the playout speed of receiver. Through simulation, we found that the combination of Token Bucket and AMP schemes provides the superiority in terms of the occurrence number of buffer underflow and overflow, the stop duration time and the number of removed frames generated by underflow and overflow, and PSNR.

• Journal of Digital Contents Society
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• v.17 no.3
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• pp.135-142
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• 2016

When video service is performed in simulation using NS-2(Network Simulation-2), the video replay is performed as the received frame order. In the existing video replay method based on the received frame order, as the frame orders of receiver and transmitter are different, the receiver buffer does not have the effect that the packets between the frames of transmitter buffer holds a regular size and packet dense and sparsity phenomenon in the receiver buffer is made by the irregular packet size due to the unpredictable reversed order of received partial frames. The above dense and sparsity phenomenon increases the probability of buffer overflow and underflow generation. To prevent these problems, the proposed frame receive order relocation method adds an extra replay buffer which rearranges the order of receive frame as the order of transmit frame, so it has the effect that the packets between the transmit frames keeps a regular size. Through the simulation using NS-2 and JSVM(Joint Scalable Video Model), the generation number of buffer overflow and underflow, and PSNR(Required Peak Signal to Noise Ratio) performance between the existing method and proposed method were compared. As a result, it was found that the proposed method would have better performance than the existing method.

• Journal of KIISE:Information Networking
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• v.28 no.4
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• pp.527-539
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• 2001

The wireless asynchronous transfer mode (ATM) system has the advantage of providing the broadband services with various quality-of-service requirements to the mobile terminal efficiently by utilizing the ATM technology developed for the wired ATM system. The available bit rate (ABR) service among various ATM services utilizes the available bandwidth remaining in the ATM link, which allows the efficient bandwidth usage. During the handoff of the mobile terminal, however, the queue length in the access point (AP) which resides in the boundary of the wired ATM network and the wireless ATM network may increase abruptly. In this paper, we propose a scheme which prevents the buffer-overflow and buffer-underflow in the AP during the handoff of the wireless ABR connection in the wireless ATM system using binary feedback rate-based ABR traffic control. This scheme controls the source's cell generation rate during both handoff period and some time interval after the completion of the handoff procedure. The simulation results show that the proposed scheme prevents the buffer-overflow and buffer-underflow. The proposed scheme can contribute to increasing the throughput of the wireless ABR service during handoff by preventing the buffer overflow and underflow during handoff period.

• Journal of KIISE
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• v.44 no.12
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• pp.1296-1306
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• 2017

Recently, as the number of Internet of Things users has increased, IETF (Internet Engineering Task Force) has released the CoAP (Constrained Application Protocol). So Internet of Things have been researched actively. However, existing studies are difficult to adapt to streaming service due to low transmission rate that result from buffer underflow. In other words, one block is transmitted one block to client's one request according to the internet environment of limited resources. The proposed scheme adaptively adjusts the rate of CON(Confirmable) message among all messages for predicting the exact network condition. Based on this, the number of blocks is determined by using buffer occupancy rate and content download rate. Therefore it improves the quality of user experience by mitigating playback interruption. Experimental results show that the proposed scheme solves the buffer underflow problem in Internet of Things streaming environment by controlling transmission rate according to the network condition.

• Proceedings of the Korean Information Science Society Conference
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• 2000.10c
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• pp.600-602
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• 2000

본 논문에서는 VOD시스템, 혹은 영상회의 시스템에서의 오디오 패킷의 QOS를 보장해 주기 위해 클라이언트의 버퍼 상태에 따른 전송속도 제어와 부가 전송 기법을 동적으로 사용하여 재전송 함으로써 오디오 패킷의 손실문제를 해결하기 위한 방법과 함께 클라이언트의 버퍼 상황을 파악하여 전송속도를 조절하여 클라이언트의 버퍼에서의 오버프로우와 언더플로우를 방지하여 VOD와 영상회의 시스템에서 오디오 데이터의 전송시 안정적인 서비스를 보장할 수 있다.

• Proceedings of the Korean Information Science Society Conference
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• 2001.04a
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• pp.250-252
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• 2001

QoS(Quality of Service)측면에서 멀티미디어 응용 서비스를 제공하기 위해서 멀티미디어 통신은 여러 가지 요구사항을 지원하여야 한다. 연속적으로 생성된 미디어의 시간적인 관계를 유지하면서 재생시킬 수 있는 동기화 메커니즘은 무엇보다도 중요한 문제이다. 전달되는 매체를 공동을 사용하기 때문에 트래픽의 폭주시, 지역적으로 멀리 떨어져 있는 경우에는 지연의 현상이 발생하게 된다. 본 논문에서는 지연에 의해 발생되는 지연지터에 대하여 완충역할을 할 수 있는 최소 버퍼크기를 제시하였다. 주문형멀티미디어서버(MOS)의 경우 각 미디어 별로 서버에 저장하여 서비스하는 연구가 현재 활발히 진행되고 있으며 이것은 네트워크의 부하를 줄일 수 있다는 장점을 가지고 있다. 또한 이러한 데이터의 경우 클라이언트에서 어느 정도 지터를 예상하여 버퍼의 크기를 갖고 있어야 하는 문제는 아직도 큰 문제점으로 나타나고 있다. 본 논문은 버퍼의 크기 및 재생에 관한 동기화 시점을 조정함으로서 버퍼의 오버플로우나 언더플로우의 현상을 방지하고자 한다. 제안한 기법은 기존의 기법보다 향상된 동기화 기법임을 시뮬레이션을 통해 검증하였다.

• KIISE Transactions on Computing Practices
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• v.21 no.12
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• pp.780-785
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• 2015

Recently HTTP (Hypertext Transfer Protocol) adaptive streaming services have been the subject of much attention. The video quality control scheme of conventional HTTP adaptive streaming services estimates bandwidth using segment throughput and smooths out the sample of segment throughput. However, the conventional scheme has the problem of QoE (Quality of experience) degradation occurring with buffer underflow and frequent quality change due to the fixed number of samples. In order to solve this problem, we propose a video quality control scheme based on segment throughput and buffer occupancy. The proposed scheme determines the number of samples according to the variation of segment throughput. The proposed scheme also controls video quality based on the threshold of bitrate to keep stable buffer occupancy. The simulation results show that proposed scheme improves QoE by preventing buffer underflow and decreasing quality change when compared with the conventional scheme.

• Proceedings of the Korea Information Processing Society Conference
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• 2001.10a
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• pp.155-158
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• 2001

QoS(Quality of Service)측면에서 멀티미디어 응용 서비스를 제공하기 위해서 멀티미디어 통신은 여러 가지 요구사항을 지원하여야 한다. 연속적으로 생성된 미디어의 시간적인 관계를 유지하면서 재생시킬 수 있는 동기화 메커니즘은 무엇보다도 중요한 문제이다. 전달되는 매체를 공동으로 사용하기 때문에 트래픽의 폭주시, 지역적으로 멀리 떨어져 있는 경우에는 지연의 현상이 발생하게된다. 본 논문에서는 지연에 의해 발생되는 지연지터에 대하여 완충역할을 할 수 있는 최소 버퍼크기를 제시하였다. 주문형멀티미디어서버(MOS)의 경우 각 미디어 별로 서버에 저장하여 서비스하는 연구가 현재 활발히 진행되고 있으며 이것은 네트워크의 부하를 줄일 수 있다는 장점을 가지고 있다. 또한 이러한 데이터의 경우 클라이언트에서 어느 정도의 지터를 예상하여 버퍼의 크기를 갖고 있어야 하는 문제는 아직도 큰 문제점으로 나타나고 있다. 본 논문은 버퍼의 크기 및 재생에 관한 동기화 시점을 조정함으로서 버퍼의 오버플로우나 언더플로우의 현상을 방지하고자 한다. 제안한 기법은 기존의 기법보다 향상된 동기화 기법임을 시뮬레이션을 통해 검증하였다.

• Proceedings of the Korean Information Science Society Conference
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• 2000.04a
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• pp.472-474
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• 2000

본 논문은 VOD 시스템에서 QoS를 보장해 주기 위한 흐름 제어 기법을 제안하고 실제 구현된 VOD 시스템에서 제안된 알고리즘을 적용하고 성능을 측정하였다. 기존의 버퍼를 관리하는 Leaky Bucket 알고리즘과 패킷의 손실률에 기반한 네트워크의 상태를 동시에 고려해서 클라이언트와 서버간에 전송 속도를 제어한다. 즉, 시간에 따라 변화하는 네트워크의 상태를 클라이너트 버퍼 관리에 반영해줌으로써 클라이언트 버퍼의 오버플로우 및 언더 플로우를 방지하고 QoS를 보장한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.10
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• pp.1976-1985
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• 2015

The amount of packets in palyout buffer of video streaming receiver can be changed by network condition, and saturated and exhausted by the delay and jitter. Especially, if the amount of incoming video traffic exceeds the maximum allowed playout buffer, buffer overflow problem can be generated. It makes the deterioration of video image and the discontinuity of playout by skip phenomenon. Also, if the incoming packets are delayed by network confusion, the stop phenomenon of video image is made by buffering due to buffer underflow problem. To solve these problems, this paper proposes the video streaming receiver with token bucket scheme which automatically establishes the important parameters like token generation rate r and bucket maximum capacity c adapting to the pattern of video packets. The simulation results using network simulator-2 (NS-2) and joint scalable video model (JSVM) show that the proposed token bucket scheme with automatic establishment parameter provides better performance than the existing token bucket scheme with manual establishment parameter in terms of the generation number of overflow and underflow, packer loss rate, and peak signal to noise ratio (PSNR) in three test video sequences.