• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.19 no.12
• /
• pp.2813-2819
• /
• 2015

In this paper, we investigate the physical layer network coding (PNC) technique in a two-way relay channel (TWRC) where two source nodes send and receive data with each other via a relay node. In particular, we consider the communication scenario where packet length from the two sources is different from each other. We analyze the bit error rate (BER) of the received packet at the relay node according to degree of overlapping between two packets. The BER of the short packet remains unchanged regardless of the degree of overlapping since the entire packet is overlapped with the longer packet, while the BER of the longer packet becomes improved as the degree of overlapping decreases. Thus, we need a novel transmission scheme to enhance BER performance of the PNC technique in TWRC environments since the overall BER performance of the PNC technique at the relay node depends on the worse BER between two ovelapping packets' BERs.

• Journal of KIISE:Information Networking
• /
• v.28 no.4
• /
• pp.540-549
• /
• 2001

Internet telephony service is very cheap and very easy to introduce the value-added service than the POTS, but is difficult to the QoS of telephone service. The existing Internet typically offers 'best effort' services only, which do not make any commitment about delay, packet loss and jitter. This paper compensates the low quality of the speech for packet loss or delay using SOLA algorithm in Internet phone. SOLA algorithm is a popular technique for Time Scale Modification of speech and audio signal. In the proposed algorithm, the receiver expands the received packet under resonable threshold, and hence compensates the QoS of speech. From the simulation, this algorithm can conceals packet loss considerably, and is also improved the quality of the speech.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.40 no.9
• /
• pp.373-383
• /
• 2003

In this paper we discuss the effect of the delay limit and the packet size related to the quality of service on a VoIP system using the Internet. We also provide a guideline to determining the optimal packet size of the voice data for a given delay limit. Empirical studies are done with two personal computers connected through the packet switched public IP network. The sender encodes the voice signal from the microphone to get PCM and ADPCM data and sends the data to the receiver using UDP packets. The receiver plays the reconstructed voice from the stream with lost and delayed packets. The quality of the reconstructed voice is evaluated offline by the MNB (Measuring Normal Block) method using the data acquired from the both sides. The result shows that under the delay limit of 100ms for 40Kbps, 32Kbps and l6Kbps of ADPCM data, the minimum packet size should be 300bytes, 400bytes and 600bytes respectively and the maximum packet size should be l200bytes commonly for the best quality of voice.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.5 no.3
• /
• pp.275-280
• /
• 2010

In this paper, VoIP(Voice over Internet Protocol) transmission performance for MANET(Mobile Ad-hoc Networks) is improved and analyzed with packet aggregation scheme which is aggregating some of short length packets to one large packet and sending to networks. VoIP simulator based on NS(Network Simulator)-2 is implemented and used to measure performance of VoIP traffic transmission. In this simulation, VoIP traffics are generated with parameters of some codes such as G.711, G.729A, GSM.AMR and iBLC. MOS(Mean Opinion Score), end-to-end network delay, packet loss rate and transmission bandwidth are measured. Performance improvements of 98% for MOS, 6.4times for end-to-end network delay, 32times for packet loss rate is shown as simulation results. On the other hand, transmission bandwidth is increased about maximum 10%. Finally, VoIP implementation guide for the performance with packet aggregation is suggested.

• The KIPS Transactions:PartC
• /
• v.12C no.3 s.99
• /
• pp.437-442
• /
• 2005

The measurement of RTT (Round Trip Time) can be used for the analysis of Internet congestion. However, simple measuring of RTT which measures only hun around time of a packet can not infer a packet forward/backward delay variation. In this thesis, we present a new algorithm which can be used for the estimation of forward/backward delay variation of packets. These delay variations are implication of network congestion state. In this algorithm, the reference forward/backward delay can be determined based on the minimum RTT value. The delay variation of each packet can be calculated by comparing reference delay with the packet delay. We verified our proposed algorithm by NS-2 simulation and delay measuring in a real network.

• The KIPS Transactions:PartA
• /
• v.13A no.7 s.104
• /
• pp.573-580
• /
• 2006

The conventional transport layer protocol TCP is designed to work under condition of packet loss is due to the network congestion, so that it's suitable in the traditional wired network with fixed hosts but it's inefficient on the wireless network where the environment of fading, noise, and transmission error comes from interference. This result from the needless transmission control of the bit error is due to treats the packet loss as a packet congestion control in the wireless network. In this paper, we propose the advanced SNOOP protocol with the consecutive packet loss and TCP window control to avoid the needless congestion management algorithm in wireless network for the wireless TCP packet transmission enhancement. We verify the performance of the advanced module from the simulation experiment result.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2004.05b
• /
• pp.475-478
• /
• 2004

In this paper, a CSSMA/AI MAC protocol for data services in packet CDMA network is presented. The proposed protocol is based on the code status sensing and reservation scheme. In the proposed protocol, the base station broadcast the rode status on a frame-by-frame basis just before the beginning of each preamble transmission, and the mobile station transmits a preamble for reserving a randomly selected code based on the received code status. After having transmitted the preamble, the mobile station listens to the downlink of the selected rode and waits for the base station reply. If this reply indicates that the code has been correctly acquired, it continues the packet transmission lot the rest of the frame. If there are other packets waiting for transmission, the base station broadcasts the status of the code as reserved, and the mobile station transmits a packet through the reserved code for the successive frames.

• Proceedings of the Korea Information Processing Society Conference
• /
• 2009.04a
• /
• pp.1190-1193
• /
• 2009

최근 무선네트워크의 급격한 발달로 이동 중에도 통신을 할 수 있는 요구사항에 IETF에서는 Mobile IPv6(MIPv6)라는 기술을 제안하였다. MIPv6는 IPv6환경에서 모바일 노드(모바일 노드)가 이동을 하면서도 통신을 할 수 있는 기술인데, 실제 상용망에서는 사용되지 않고 있다. 그 이유는 모바일 노드에 이동성을 제공하는 모바일 스택의 탑재가 높은 오버헤드를 가지고 있고, 시그널링 비용이 많이 들기때문이다. 이와 같은 문제를 해결하기 위해서 지난 해 Proxy Mobile IPv6(PMIPv6)가 표준화 되었다. PMIPv6는 모바일 노드에 이동성을 제공하는 모바일 스택을 올리지 않고 네트워크 엔티티가 대신 모바일 스택을 가지고 시그널링을 처리하기 때문에 결과적으로 시그널링 비용을 줄였다. 하지만 PMIPv6에서는 인접한 네트워크 사이에서 패킷을 전송할 때 MAG간의 통신이 가능하지 않고, 오직 LMA를 통하여 대응 노드(대응 노드)에게 패킷을 전송할 수 있다. 게다가 같은 서브넷에 있는 대응 노드에게 패킷을 전송할 때도 LMA를 거치게 되는 불필요한 과정이 발생한다. 게다가 LMA를 통하여 패킷을 전송하는 방식은 지연과 리오더링(Re-Ordering)을 발생시켜본 패킷 재전송을 유발하여 전송 품질을 떨어뜨리게 된다. 본 논문에서는 인접한 MAG간의 통신, 또는 같은 서브넷 내에서의 통신에서 LMA를 거치지 않고 통신을 할 수 있는 방안으로 블룸 필터를 이용한 통신 기법을 제안한다.

• Journal of KIISE:Information Networking
• /
• v.27 no.3
• /
• pp.315-322
• /
• 2000

네트워크에서 QoS를 보장하기 위해 최근에 제안되는 패킷 스케줄링 알고리즘은 대부분 우선 순위에 입각한 패킷 전송 서비스를 한다. 이러한 우선 순위를 유지하기 위한 큐의 관리에는 많은 비용이 들므로 QoS를 보장하는 네트워크에서 우선 순위 큐의 관리 비용을 줄이는 노력이 필요하다. 패킷 스케줄링 알고리즘 중 RPO+(Rotate Priority Queue)는 우선 순위 FIFO(First in first out)큐를 사용하여 주기 적으로 재명명되는 패킷 스케줄링 알고리즘이다. FIFO 큐에 패킷들을 근사 정렬하여 패킷의 우선 순위를 유지하므로 계산 복잡도를 줄이지만, 패킷 우선 순위를 유지하기 위해 2배(2P)의 큐를 필요로 한다.[1] 본 논문에서는 필요한 큐의 개수를 P개의 큐로 제한하여 큐에 대한 관리 비용을 줄였으며 엔터프라이즈 환경에서 애플리케이션이 요구하는 서비스 특성에 따라 클래스로 구분하여 적합한 패킷 스케줄링 서비스를 제공하는 알고리즘을 제시한다. 본 기법은 추가적인 오버플로우 큐를 관리하고 패킷 어드미션 컨트롤러를 통해 패킷 전송 지연 시간을 제한함으로 다양한 애플리케이션의 네트워크 QoS 요구를 보장하고 패킷 전손 효율을 높였다.

• 한국ITS학회:학술대회논문집
• /
• v.2006 no.10
• /
• pp.199-202
• /
• 2006

본 논문에서는 IPv6를 이용한 새로운 T-DMB 멀티미디어 서비스 시스템을 설계 및 구현하였다. 본 논문에서 제안한 방식은 IPv6의 듀얼스택 기술인 DSTM 방식을 T-DMB 패킷 전송 서비스에 접목시켰다. IPv6의 주소 체계 방식을 이용하여 기존의 T-DMB와의 호환성을 유지하면서 T-DMB 단말 내에 DSTM 패킷 전송을 위한 어플리케이션 데이터 테이블을 배치하여 T-DMB in IPv6 터널링을 통한 서비스를 실행할 수 있게 하였다. 또한, T-DMB Client 와 DSTM TEP를 단말에 배치하여 패킷 전송을 가능하게 하였다. T-DMB in IPv6 터널링 기술을 도입하여 T-DMB 방송 시스템의 성능 평가를 실시한 결과, 기존의 T-DMB 방송 서비스보다 약 2배 정도의 전송 용량 증가 및 채널 용량 증가가 효과를 바로 볼 수 있었고, 패킷 크기 따른 전송 효과는 기존의 방송 서비스보다 약 3%정도 증가된 결과 값을 얻을 수 있었다.