• Journal of Digital Contents Society
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• v.17 no.3
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• pp.181-188
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• 2016

Video streaming traffics which are arrived in receiver have irregular traffic patterns by many problems over the network path. Particularly, if these received traffics enter into replay buffer without any operation, the overflow and underflow effects are made according to the buffer status. There was an existing scheme which automatically set up token bucket parameters using the video information file under the assumption of the lossless packet on network. The existing scheme has a problem which can set up the wrong token bucket parameters by the lost packets on the practical networks with video packet loss. Therefore, this paper proposes a new scheme which reset up video file information to correct the wrong token bucket parameters in case of packet loss in practical networks with packet loss. Through the simulation, it was found that the proposed scheme would have better performance than the existing scheme in terms of overflow generation and packet loss.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.8A
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• pp.858-870
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• 2007

TCP is popular protocol for reliable data delivery in the Internet. In recent years, wireless environments with transmission errors are becoming more common. Therefore, there is significant interest in using TCP over wireless links. Previous works have shown that, unless the protocol is modified, TCP may perform poorly on paths that include a wireless link subject to transmission errors. The reason for this is the implicit assumption in TCP that all packet losses are due to congestion which causes unnecessary reduction of transmission rate when the cause of packet losses are wireless transmission errors. In this paper, we propose a new LDA that monitors the network congestion level using ROTT. And we evaluate the performance of our scheme and compare with TCP Veno, Spike scheme with NS2(Network Simulator 2). In the result of our experiment, our scheme reduces the packet loss misclassification to maximum 55% of other schemes. And the results of another simulation show that our scheme raise its transmission rate with the fairness preserved.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1998.05a
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• pp.270-275
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• 1998

무선망에서는 높은 에러율과 핸드오프로 인해 패킷 손실이 자주 발생한다. 그러나, 기존의 TCP/IP 프로토콜에서는 패킷 손실을 혼잡에 의해서 발생한 것이라 보고, 패킷 송신율을 줄임으로써 혼잡제어를 실시한다. 따라서 기존의 TCP/IP 프로토콜을 무선망에 그대로 적용했을 경우 매우 낮은 성능을 나타낸다. 본 연구에서는 무선망에서의 국부적인 재전송을 통하여 패킷 손실을 보완하고, 사용자가 셀 간을 이동할 때 핸드오프 지연시간을 줄일 수 있는 새로운 핸드오프 방식을 제안한다. 시뮬레이션을 통해서 제안된 방법을 성능을 평가한다.

• Proceedings of the Korea Multimedia Society Conference
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• 2002.05c
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• pp.416-421
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• 2002

본 논문에서는 인터넷 상에서 UDP 소켓 인터페이스를 이용하여 실시간 스트리밍 전송을 시뮬레이션 하였고 또한 수신자로부터 피드백 된 정보를 이용하여 송신자가 전송률을 제어하는 것을 시뮬레이션 하였다. 수신자로부터 제어정보를 피드백 받아 송신자가 전송률을 제어하는 경우와 송신자가 전혀 전송률제어를 하지 않는 경우의 패킷 손실 값을 비교했다. 결과로서, 피드백 되는 정보를 이용하여 제어를 하는 경우가 지속적인 패킷손실을 방지함을 보여 준다.

• Journal of KIISE:Information Networking
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• v.32 no.3
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• pp.382-390
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• 2005

As today's networks evolve towards an If-based integrated network, the role of transmission control protocol(TCP) has been increasing as well. As a well-known issue, the performance of TCP is affected by its loss recovery mechanism that is comprised of two algorithms; fast retransmit and fast recovery. Although retransmission timeout(RTO) caused by multiple packet losses can be avoided by using selective acknowledgement(SACK) option, RTO cannot be avoided if a retransmitted packet is lost. Therefore, we propose a simple modification to make it possible for a TCP sender using SACK option to detect a lost retransmission. In order to evaluate the proposed algorithm, simulations have been performed for two scenarios where packet losses are random and correlated. Simulation results show that the proposed algorithm can improve TCP performance significantly.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.8C
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• pp.810-823
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• 2005

The transmission of duplicate packets provides a great loss-resilience without undue time-delay in the video transmission over packet loss networks. But this method generally deteriorates the problem of traffic congestion because of the increased bit-rate required for duplicate transmission. In this paper, we propose an efficient packetization and duplicate transmission of video packets. The proposed method transmits only the video signal with high priority for each video macroblock that is quite small in volume but very important for the reconstruction of the video. The proposed method significantly reduces the required bit-rate for duplicate transmission. An efficient packetization method is also proposed to reduce additional packet overhead which is required for transmitting the duplicate data. The duplicated high priority data of the Previous video slice is transmitted as a Piggyback to the data Packet of the current video slice. It is shown by simulations that the proposed method remarkably improves the packet loss-resilience for video transmission only with small increase of redundant duplicated data for each slice.

• Journal of Korea Multimedia Society
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• v.6 no.7
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• pp.1231-1238
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• 2003

As the demand on multicast services increases to support the terminal mobility in the midst of the development of wireless mobile communication technology, many new methods have been studied. However, previous methods occur delay and packet loss during re-joining to multicast service group when moving mobile node. In this paper, we will propose a scheme that decreases packet loss using regional registration method. Proposed scheme modified signaling message of previous regional-registration and added caching. In case move to same GFA, we can prevent packet loss by keeping registration to FA until perfectly registration ends. Also, In case move to other GFA, we can reduce packet loss by using cache. Therefore, we can receive efficient multicast service.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2010.07a
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• pp.126-129
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• 2010

본 논문에서는 패킷 오류가 발생하는 IP 망을 통해 비디오 전송 서비스를 제공하기 위한 조건적 ARQ 기반의 오류 제어기법을 제안한다. IP 망에서 패킷 손실이 발생하면 수신 단말은 서버에게 손실된 패킷의 재전송을 요청할 수 있다. 그러나 망 상황과 접속 단말 개수를 고려하지 않고 무조건적인 재전송을 요청 할 경우 손실된 패킷이 복호화에 필요한 시점까지 도착 할 것인지가 보장되지 않을 뿐더러, 오히려 불필요한 재전송 요청에 의해 망에서의 트래픽 량을 가중시킬 수 있다. 따라서 망 상황을 고려하여 손실된 패킷이 재전송에 의해 적절한 시간 내에 도착이 가능한지를 판단하여 조건적으로 ARQ를 요청하는 기술이 요구된다. 본 논문에서는 기존의 RFC 문서에서 권고하는 재전송 관련 메카니즘들을 효과적으로 활용하여 표준기술 기반의 조건적 ARQ 오류제어 기법을 제안한다. 제안된 기법에서는 단말에서 추정된 망 상황을 기반으로 조건적으로 재전송 요청 여부를 결정하고, 더 나아가서 유효한 재전송 요청 시점을 결정하여 보다 효율적인 ARQ 가 적용될 수 있도록 한다. 제안된 조건적 ARQ 기반의 오류 제어 기법의 성능을 검증하기 위하여 패킷 손실 네트워크 환경을 NIST-Net을 활용하여 구축하며 실험 결과를 통해 제안된 기법의 우수한 오류 강인 성능을 확인한다.

• Journal of Internet Computing and Services
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• v.6 no.4
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• pp.115-124
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• 2005

The packet loss problem that occurs in the mobile multicast (MoM) protocol due to designated multicast service provider(DMSP) handoff is investigated through simulation experiments for several DMSP selection policies. Then, two enhanced DMSP schemes are proposed to minimize the packet loss of the MoM protocol with single DMSP. The first scheme uses a backup DMSP and greatly reduces the packet loss rate at the expense of the increased network traffic. The second scheme utilizes the extended DMSP operation and shows many desirable features such as the almost zero packet loss rate and relatively low network traffic.

• Proceedings of the Korean Information Science Society Conference
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• 2012.06d
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• pp.346-347
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• 2012

네트워크의 안정성을 보장하면서 멀티미디어 스트리밍 서비스의 품질을 향상시키기 위해서 TFRC(TCP-Friendly Rate Control) 프로토콜이 제안되었다. 그러나 TFRC의 Slow-start 알고리즘은 TCP와 같이 가용대역폭을 초과하여 패킷을 전송하는 오버슈트 (Overshoot) 문제로 인해 스트리밍 비디오 화질을 열화시키는 패킷 손실을 발생시킨다. 본 논문에서는 TFRC의 오버슈트에 의한 패킷 손실을 줄이기 위한 Slow-start 알고리즘을 제안한다. 제안하는 알고리즘은 네트워크의 혼잡 정도에 따라서 전송률의 증가량을 감소시킴으로써 패킷 손실을 감소시켰다. 시뮬레이션을 통해서 제안하는 알고리즘이 기존의 Slowstart 알고리즘 보다 적은 패킷 손실을 발생시키는 것을 보였다.