A WAP-Based Session Layer Supporting Distributed Applications in Nomadic Environments

위성 통신은 넓은 지역에 통신 서비스를 제공하는 광역 네트워크이다. 하지만, 위성 통신은 대역폭이 제한되고 매우 긴 전파 지연 시간과 높은 BER(Bit Error Rate)로 인해 위성 통신을 고려하지 않은 TCP의 성능이 저하 된다. 본 논문에서는 위성 통신에서의 TCP 성능을 향상시키기 위해 DVB-RCS(Digital Video Broadcasting-Return Channel via Satellite) 네트워크 환경을 고려한 cross-layer 기반 PEP(Performance Enhancing Proxy) 기술을 제안한다. 제안하는 프로토콜은 TCP와 링크 계층 간 위성 자원 할당 정보를 정보 교환을 통해 최적의 TCP CWND(Congestion Window)를 설정한다. 제안한 프로토콜의 성능 평가를 위해 리눅스 기반의 PEP 테스트 베드를 구현하였다. 성능 평가 결과 다양한 BER에서 제안한 프로토콜은 위성 자원 할당 정보를 이용하여 최적의 TCP CWND 크기를 설정하기 때문에 단일 및 다중 세션 환경에서 기존 TCP 보다 제안한 프로토콜이 더 좋은 성능을 보여준다는 것을 확인할 수 있었다. Satellite communication is a wide area network (WAN) which provides communication service worldwide. However, the performance of TCP can be seriously degraded in the satellite networks due to limited bandwidth, long round-trip time (RTT) and high bit error rate (BER) over satellite links. In order to improve the performance of TCP, this paper proposes cross-layer Performance Enhancing Proxy (PEP) in digital video broadcasting-return channel via satellite (DVB-RCS) networks. The proposed protocol sets TCP Congestion Window (CWND) size by using satellite resource allocation information exchanged between TCP and the link-layer. we implement PEP testbed based on Linux to evaluate the performance of the proposed protocol. The simulation results show that the proposed protocol performs better than standard TCP both in single and multiple sessions in variant BER, because the proposed protocol sets TCP CWND size by using satellite resource allocation.

Currently there is no control for real-time traffic sources in IP networks. This is a serious problem because real-time traffic can not only congest the network but can also cause unfairness and starvation of TCP traffic. A new rate control scheme, RCS, is introduced for real-time traffic in networks with high bandwidth-delay products and high bit error rates. RCS is based on the concept of using dummy packets to probe the availability of network resources. Dummy packets are treated as low priority packets and consequently they do not affect the throughput of actual data traffic. Therefore, RCS requires all the routers in the connection path to support some priority policy. Simulation experiments show that in environments with high bandwidth-delay products and high bit error rates, RCS achieves good throughput performance without penalizing TCP connections.

As the Internet dramatically changes the world, applications that based on TCP/IP are taking up an increasing proportion in satellite services. Since there are characteristics such as long round trip delay and high bit error rate in satellite communication links, when combined with TCP/IP services, it will lead to frequently data resending and decrease network throughput. An accelerator based on Space Communication Protocol Standard Transport Protocol (SCPS-TP) has been designed and presented in this paper functioning as a Performance Enhancing Proxy (PEP) to improve TCP performance in satellite IP networks. Testing results show that satellite IP networks using the accelerator can achieve higher throughput performance, shorter transfer delay, better stability and stronger flexibility. The accelerator is totally transparent to the applications, and therefore it could be applied to any kind of space communication, and it could also meet the demand for rapid response in military, earthquakes and other emergencies.

This Satellite network has many disadvantages such as long round-trip-time (RTT), high bit error rate. The issues above pose challenge to existing Slow start algorithm. Slow start algorithm is a key component which directly affects startup performance of TCP. This paper mainly improves startup performance of TCP and proposes a slow start algorithm that adopts logarithmic function to overcome multiple congestion packet loss caused by inaccurate ssthresh (slow start threshold) estimation and improves throughput performance.

Satellite communication has become important in the current global data communication due to its large coverage area. Satellite links are affected by two major concerns: long propagation delays, and relatively high error rates. Standard TCP congestion control variants were not designed to address these challenges. They generally depend on round trip times (RTTs), and hence suffer heavily if the RTTs are large. Various TCP variants like Hybla have been proposed in order to optimize the channel throughput in heterogeneous networks incorporating satellite links. The error rate of satellite channels can be substantially mitigated using forward error correction (FEC) techniques. In such a situation, according to the results obtained by our comparative study, the performance of Cubic, which is designed for high-speed networks, is better than Hybla, in conditions of high-latency and low error rates.

Modified Packet Combining (MPC) scheme is an error correction scheme, which can correct double or, higher bit errors. In some earlier work, a scheme known as Packet Reversed Packet Combining (PRPC) scheme that will correct error if occurs at same bit location of erroneous copies, was studied. Both PRPC and MPC scheme are believed to offer higher throughput and better error correction capability at high bit error rate (BER) and large packet size only. Exact analysis was performed to substantiate this claim of higher throughput and better error correction capability in high as well as in low bit error rate (BER) for large and medium packet size. To combat high bit error rate some researches proposed to use Error Correction Code (ECC) in place of Error Detection Code (EDC) in ARQ scheme. But no conclusive result is found to justify the idea in terms of throughput. The current investigation is an attempt to explore the basic gap. In this paper, we investigate the exact analysis of both PRPC and MPC in a Modified way over Conventional PRPC, and report the findings thereof. We propose a combined protocol of Modified PRPC and MPC, and find that it will offer higher throughput and better error correction capability at high and low BER with medium and large packet size as we have used MPC (Modified Packet Combining) scheme with PRPC (Packet Reverse Packet Combining) scheme in a Modified way by combining BEC (Backward Error Correction) with FEC (Forward Error Correction) Codes for Mobile Communication and Networks.

REDUP: a packet loss recovery scheme for real-time audio streaming over wireless IP networks

Transmission Control Protocol (TCP) is known to suffer from performance degradation in mobile wireless environments, as such environments are prone to packet losses due to high bit error rates and mobility induced disconnections, because TCP is well reliable protocol for wired networks. In wired network packet loss due to congestion. Proposed scheme is a true end-to-end approach, based on the idea of exclusive handover message and is used for alleviating the degrading effect of host mobility on TCP performance. Experiments are performed using the Network Simulator (NS-2). The simulator has been extended to incorporate wireless link characteristics.

Timer expiration times shorter than round-trip time are shown to possibly yield superior efficiency of selective repeat protocols, especially in the case of higher delay-bandwidth products and higher bit error rates. Both three-layer and two-layer communication system models are considered. A new timeout management strategy is proposed to be used in a two-layer system allowing a safe decrease of timer expiration time without the risk of inter-layer congestion. Simulation results confirm expectations, and the proposed strategy is compared with previously known multiple-copy transmission methods.

In terrestrial-satellite hybrid networks, traditional TCP performance degrades a lot due to long propagation delays and high bit error rates link. Performance enhancing proxy(PEP) can shorten round trip time loop time and Peach can enhance the performance over high bit error rates link. In this paper, we propose a new scheme that combines PEP and TCP Peach for terrestrial-satellite hybrid networks. The scheme could reduce the negative impact on performance (in terms of throughput). To obtain an analysis result, we also introduce a mathematic model for our PEP. The analysis and simulation results demonstrate that our scheme could significantly improve the network performance than traditional TCP protocols.

WiMAX technology has emerged as an exciting technology for next generation broadband wireless network with promises to offer high throughput and long coverage. WiMAX, however, has yet to prove its merits when the wireless nodes are not fixed and move at high vehicular speeds. Recent studies suggest that WiMAX offers limited throughput when a mobile node travels at vehicular speeds. Multipath fading causes high bit error rates at the receiver and is considered the main reason behind low throughput at high vehicular speeds. Bit error rates and maximum packet size govern the packet error rates, and error recovery for higher number of corrupted packets is not an attractive option for many real-time applications with tight delay and jitter constraints. In this paper, we present a mathematical model for estimating bit error rates in WiMAX communication at vehicular speeds. The estimated bit error probability is taken into account for proactively computing an optimum packet size that offers the best chance of achieving improved throughput. We further propose a model that can be used to increase the utility of real-time wireless applications (e.g., video surveillance systems in public train) based on the knowledge of estimated transmission rates. We simulated the proposed and other standard schemes for a centralized video surveillance system in a public train where the train moves at high speeds and sends real-time video data to a central control room through wireless channels. The results show that the proposed scheme achieves significantly higher throughput, lower jitter and higher utility gain compared to the standard schemes.

Different from terrestrial sensor networks, underwater sensor networks use acoustic channels for communication and most sensor nodes are mobile due to water current. These distinctions feature underwater sensor networks with highly dynamic topology, high bit error rate, large propagation delay, and low available bandwidth. Traditional ARQ series are not suitable due to their frequent feedbacks, which is undesirable in underwater environment. On the other hand, pure FEC can't apply into underwater sensor networks, not only because of its complexity, but also because it needs some feedbacks. This paper studied transport protocol in underwater sensor networks, implemented a reliable lightweight RFEC-based encoding and decoding algorithm. Then combine the algorithm with ARQ to a transport protocol. In this way, take advantages of FEC to deal with high bit error, and ARQ to avoid feedbacks. Through simulation experiment on method of random sequences generation, parameters selection, and transmission strategy, some useful conclusion is gotten. It provides practical data and references on practical application of underwater sensor networks.

Near-threshold voltage (NTV) operation is expected to enable up to 10× energy-efficiency for future processors. However, reliable operation below a minimum voltage (Vccmin) cannot be guaranteed. Specifically, SRAM bit-cell error rates are expected to rise steeply since their margins can easily be violated at near-threshold voltages. Multicore processors rely on fast private L1 caches to exploit data locality and achieve high performance. In the presence of high bit-cell error rates, an L1 cache can either sacrifice capacity or incur additional latency to correct the errors. We observe that L1 cache sensitivity to hit latency offers a design tradeoff between capacity and latency. When error rate is high at extreme Vccmin, it is worthwhile incurring additional latency to recover and utilize the additional L1 cache capacity. However, at low error rates, the additional constant latency to recover cache capacity degrades performance. With this tradeoff in mind, we propose a novel private L1 cache architecture that dynamically learns and adapts by either recovering cache capacity at the cost of additional latency overhead, or operate at lower capacity while utilizing the benefits of optimal hit latency. Using simulations of a 64-core multicore, we demonstrate that our adaptive L1 cache architecture performs better than both individual schemes at low and high error rates (i.e., various NTV conditions).

In order to solve the problems of the traditional anti-jamming algorithm of broadband wireless communication, such as poor anti-jamming performance and high bit error rate, an anti-jamming algorithm of broadband wireless communication based on Embedded MCU is proposed. In the broadband wireless communication based on embedded single-chip microcomputer, the m-sequence of communication signal data is constructed, encoded and decoded. Finally, CRC redundancy test and error correction are carried out for the decoded communication data, so far the design of broadband wireless communication anti-interference algorithm based on embedded single-chip microcomputer is completed. Through the contrast experiment, compared with the traditional anti-jamming algorithm of broadband wireless communication, the experimental results show that compared with the traditional anti-jamming algorithm of broadband wireless communication, the proposed anti-jamming algorithm of broadband wireless communication based on Embedded MCU has lower bit error rate, which shows that it has better anti-jamming ability.

While Transmission Control Protocol (TCP) works well with a low bit error rate (BER), the performance of TCP degrades significantly if the BER rises above a certain level. A study of the performance of TCP with high BER is required for the efficient design and deployment of such systems. In this paper, we address the problem of TCP performance in high BERs and analyze the issues by investigating the effect of BERs on system performance. We consider TCP Reno in our study to explore the system performance using extensive analysis of simulation and modeling. In the analysis, we consider the amount of datagram sent and retransmitted, mean throughput, link-layer overhead, TCP window size, FTP download response time, packet dropping and retransmission, and the TCP congestion avoidance mechanism. We validate simulation results by setting up a virtualized testbed using Linux hosts and a Linux router. The results obtained show that TCP throughput degrades significantly and eventually collapses at the packet drop probability of 10% (BER = 10−5). The FTP download response time is about 32 times longer than that of a perfect channel (no packet dropping). We found that TCP Reno cannot handle such a high BER to operate in wireless environments effectively. Finally, we provide recommendations for network researchers and engineers confronted with the challenge of operating TCP over noisy channels.