• 한국정보통신학회논문지
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• 제25권4호
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• pp.581-587
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• 2021

실시간 비디오 스트리밍은 컨텐츠 전달 및 원격회의 등에 사용되는 기술로, 전송지연에 민감한 기술 중 하나이다. 최근 코로나바이러스 감염증(COVID-19)으로 인해 개인방송/원격회의와 같은 개인이 제작 및 이용하는 컨텐츠들이 증가하면서, 실시간 비디오 스트리밍 시 전송지연 저감과 더불어 스트리밍 되는 컨텐츠에 대한 보호의 필요성이 강조되고 있다. 따라서, 본 연구에서는 실시간 비디오 스트리밍에 선별적 비디오 암호화 방법을 적용할 경우 발생하는 전송지연을 저감하기 위한 패킷 응집 관련 알고리즘을 제안하였다. 제안 방법은 선별적 비디오 암호화 프레임워크에서 패킷 단위로 재배치된 비디오 데이터 전송 시 한 번에 전송되는 크기의 조절이 가능하도록 개선하였으며, 실제 테스트베드를 통한 실험결과는 제안 방법 적용 시 기존대비 전송지연을 약 11% 저감할 수 있음을 보여주었다.

• 대한임베디드공학회논문지
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• 제8권6호
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• pp.311-318
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• 2013

In this paper, we consider a new packet scheduling algorithm for real-time traffic in the HSDPA system that has been introduced for the WCDMA system, in order to provide high transmission rates. The objective of the design is to meet the maximum tolerable delay and consider channel assignment based on the received SIR for real-time traffic users. The proposed scheduling algorithm shows that the users are ranked by the ratios of the bits in the buffer to the residual time for transmission as priority order; then the ranked users are assigned certain number of channels based on the SIR value table. The simulation results show that the proposed algorithm can provide a lower packet drop rate, and satisfy real time quality of service (QoS) requirements.

• 한국정보통신학회논문지
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• 제14권5호
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• pp.1311-1321
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• 2010

주변환경에서 정보를 수집하는 센서네트워크는 다양한 분야에서 사용되고 있다. 정밀한 분석을 위하여 매우 짧은 주기로 정보를 수집할 필요가 있는 어플리케이션에서는 네트워크를 구성하는 노드의 수가 증가하고 정보를 수집하는 주기가 짧아질수록 생성되는 데이터의 양도 증가하기 때문에 노드 간에 지연을 최소화하여 효율성을 높일 필요가 있다. 본 논문에서는 네트워크 구성시 데이터 전송률에 영향을 미치는 전송주기와 센서 노드간의 거리, 전송 패킷의 크기에 대한 실험을 실시하여 실험한 결과를 바탕으로 주기적 전송 기법의 한계와 문제점을 분석하고 패킷 전송 작업이 완료되었음을 알려주는 전송 완료 이벤트를 이용하여 연속적인 패킷 전송시 지연을 줄여주는 SET(SendDone Eventbased Transmission Technique)기법을 제안한다. SET에서는 패킷 전송이 완료된 시점에 즉시 다음 패킷을 전송하기 때문에 패킷을 전송하는데 소요되는 시간에 상관없이 패킷 전송간에 지연이 발생하지 않는다. 따라서 연속적으로 대량의 패킷을 전송할 때 높은 전송율을 제공할 수 있다.

• 한국멀티미디어학회논문지
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• 제18권4호
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• pp.492-498
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• 2015

In general, data packets from sensor nodes are transferred to the sink node in a wireless sensor networks. So many data packets are gathered around the sink node, resulting in significant packet collision and delay. In this paper, we propose an efficient packet transmission mechanism for multi-hop wireless sensor networks. The proposed mechanism is composed of two modes. One mode works between sink node and 1-hop nodes from sink. In this mode, data packets are transmitted in predefined time slots to reduce collisions. The other mode works between other nodes except sink node. In this mode, duplicated packets from neighbor nodes can be detected and dropped using some control signals. Our numerical analysis and simulation results show that our mechanism outperforms X-MAC and RI-MAC in terms of energy consumption and transmission delay.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅲ
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• pp.1629-1632
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• 2003

Generally an increasing number of recent applications have found TCP too limiting. There are some characteristics in the transmission of document and binary data which some transmission delay are tolerant but the content must completely be transferred. However voice signals are more sensitive with not some packet loss but some transmission delay. Therefore, Stream Control Transmission Protocol(SCTP) is proposed to minimize the delay and packet loss in the field of delivery of voice signal. SCTP is designed to transport PSTN signalling messages over IP networks, but is capable of broader applications. In this paper, the architecture of SCTP implementation is designed and some interface of SCTP software library which are implemented are specified.

• 한국통신학회논문지
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• 제13권2호
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• pp.171-181
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• 1988

본 논문에서는 packet-switched network에서 음성을 서비스하는데 있어서 고려해야 할 여러가지 점들을 살펴보고, 실제로 음성과 데이터를 동시에 서비스하는 packet voice/data terminal을 구현하였으며 그 성능 분석을 시도하였다. PVDT의 software는 OSI 7 layer architecture에 맞추어 설계하였으며 음성과 데이터를 link level부터 구별하여 서비스하였다. 또한 음성 packet의 전송 delay를 작게 하기 위해 데이터보다 음성을 우선적으로 서비스하도록 하였으며 간략화된 protocol로 재전송에 의한 overhead를 없앴다. PVDT의 hardware의 구성은 기능별로 master control module, speech processing module, speech activity detection module, telelphone interface module, input/output inteface module로 나누어진다. Packet음성통신망에 대한 해석으로는 음성 packet의 전송 delay의 variance에 의한 영향을 줄이기 위한 최적 재생지연시간을 전송 delay의 분포를 통해 계산하였다.

• ETRI Journal
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• 제34권6호
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• pp.807-815
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• 2012

In this paper, we propose a scheme for partially dynamic lane control for energy saving in multilane-based high-speed Ethernet. In this scheme, among the given transmission lanes, at least one lane is always operating, and the remaining lanes are dynamically activated to alleviate the network performance in terms of queuing delay and packet loss in the range of acceptance. The number of active lanes is determined by the decision algorithm based on the information regarding traffic and queue status. The reconciliation sublayer adjusts the transmission lane with the updated number of lanes received from the algorithm, which guarantees no processing delay in the media access control layer, no overhead, and minimal delay of the exchanging control frames. The proposed scheme is simulated in terms of queuing delay, packet loss rate, lane changes, and energy saving using an OPNET simulator. Our results indicate that energy savings of around 55% (or, when the offered load is less than 0.25, a significant additional savings of up to 75%) can be obtained with a queuing delay of less than 1 ms, a packet loss of less than $10^{-4}$, and a control packet exchange time of less than $0.5{\mu}s$ in random traffic.

• 전자공학회논문지S
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• 제35S권3호
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• pp.65-75
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• 1998

We consider connection-oriented wireless cellular networks such as the second generation wireless cellular networks and wirelss ATM networks. In these networks, a separate forward signaling channel is provided for the transmission of paging and channel allocation packets. When a call destined to a user is requested, all the base stations in the user's current location area broadcast the corresponding paging packet across forward signaling channels. By slot mode operation and paging group allocation for fusers in a location area, we can reduce relative power consumption level at battery-operated terminals. However, a sthe number of paging groups is increased for lowering relative power consumption level, a paging packet experiences higher delay to access the forward signaling channel. For the pre-negotiated quality-of-service level, paging packet delay level must be limited. In this paper, we consider static and dynamic multiplexing schemes for paging packets, and develop an analytical method for calculating paging packet delay and relative power consumption levels. Using this analytial method, we investigate the effect of network parameters on the paging packet delay and relative power consumption levels.

• 한국경영과학회:학술대회논문집
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• 대한산업공학회/한국경영과학회 1995년도 춘계공동학술대회논문집; 전남대학교; 28-29 Apr. 1995
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• pp.729-738
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• 1995

We consider a priority-based packet-switching system with three phases of the packet transmission time. Each packet belongs to one of several priority classes, and the packets of each class arrive at a switch in a Poison process. The switch transmits queued packets on a priority basis with three phases of preemption mechanism. Namely, the transmission time of each packet consists of a preemptive-repeat part for the header, a preemptive-resume part for the information field, and a nonpreemptive part for the trailer. By an exact analysis of the associated queueing model, we obtain the Laplace-Stieltjes transform of the distribution function for the delay, i.e., the time from arrival to transmission completion, of a packet for each class. We derive a set of equations that calculates the mean response time for each class recursively. Based on this result, we plot the numerical values of the mean response times for several parameter settings. The probability generating function and the mean for the number of packets of each class present in the system at an arbitrary time are also given.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제6권4호
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• pp.1006-1025
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• 2012

Improving the quality of service in IP networks is a major challenge for real-time voice communications. In particular, packet arrival-delay variation, so-called "jitter," is one of the main factors that degrade the quality of voice in mobile devices with the voice-over Internet protocol (VoIP). To resolve this issue, a receiver-based enhanced timing recovery algorithm combined with active jitter estimation is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. Extensive simulations have shown that the proposed algorithm delivers high voice quality by pursuing an optimal trade-off between average buffering delay and packet loss rate.