Forward Error Correction Mechanism Research Articles

Packet Loss Rate Prediction Using the Sparse Basis Prediction Model

The quality of multimedia communicated through the Internet is highly sensitive to packet loss. In this letter, we develop a time-series prediction model for the end-to-end packet loss rate (PLR). The estimate of the PLR is needed in several transmission control mechanisms such as the TCP-friendly congestion control mechanism for UDP traffic. In addition, it is needed to estimate the amount of redundancy for the forward error correction (FEC) mechanism. An accurate prediction would therefore be very valuable. We used a relatively novel prediction model called sparse basis prediction model. It is an adaptive nonlinear prediction approach, whereby a very large dictionary of possible inputs are extracted from the time series (for example, through moving averages, some nonlinear transformations, etc.). Only few of the very best inputs among the dictionary are selected and are combined linearly. An algorithm adaptively updates the input selection (as well as updates the weights) each time a new time sample arrives in a computationally efficient way. Simulation experiments indicate significantly better prediction performance for the sparse basis approach, as compared to other traditional nonlinear approaches.

Generic forward error correction of short frames for IP streaming applications

If the frame size of a multimedia encoder is small, Internet Protocol (IP) streaming applications need to pack many encoded media frames in each Real-time Transport Protocol (RTP) packet to avoid unnecessary header overhead. The generic forward error correction (FEC) mechanisms proposed in the literature for RTP transmission do not perform optimally in terms of stability when the RTP payload consists of several individual data elements of equal priority. In this paper, we present a novel approach for generating FEC packets optimized for applications packing multiple individually decodable media frames in each RTP payload. In the proposed method, a set of frames and its corresponding FEC data are spread among multiple packets so that the experienced frame loss rate does not vary greatly under different packet loss patterns. We verify the performance improvement gained against traditional generic FEC by analyzing and comparing the variance of the residual frame loss rate in the proposed packetization scheme and in the baseline generic FEC.

Adaptive QoS Routing by Cross-Layer Cooperation in Ad Hoc Networks

QoS provisioning is a complex and challenging issue in mobile ad hoc networks, especially when there are multiple QoS constraints. In this paper, we propose an adaptive QoS routing scheme supported by cross-layer cooperation in ad hoc networks. The cross-layer mechanism provides up-to-date local QoS information for the adaptive routing algorithm, by considering the impacts of node mobility and lower-layer link performance. The multiple QoS requirements are satisfied by adaptively using forward error correction and multipath routing mechanisms, based on the current network status. The complete routing mechanism includes three parts: (1) a modified dynamic source routing algorithm that handles route discovery and the collection of QoS-related parameters; (2) a local statistical computation and link monitoring function located in each node; and (3) an integrated decision-making system to calculate the number of routing paths, coding parity length, and traffic distribution rates. Simulation results are presented to illustrate the overall performance of our scheme. Our results indicate that our adaptive routing scheme provides suitable QoS performance that is less sensitive to network conditions (i.e., node mobility, transmission power, channel characteristics, and the traffic pattern) than a nonadaptive routing strategy.

이동 무선 네트워크의 전송 성능 향상을 위한 적응적 FEC 알고리즘

무선 이동 네트워크에서 대부분의 패킷 손실은 네트워크 체증(congestion)보다는 전송(propagation) 오류에 의해 발생된다. 이러한 전송 오류에 의한 패킷 손실을 방지하여 전송 효율을 향상하기 위해 무선 네트워크에서는 FEC(Forward Error Correction)알고리즘을 채택하고 있다. 그러나 정적인 FEC방식은 연속적으로 변화하는 무선 채널의 전송 오류율에 알맞은 정정 코드(check code)를 채택하지 못해 도리어 전송 효율이 저하되는 문제가 있다. 이러한 문제를 개선하기 위해서는 채널의 상태에 따라 FEC를 동적으로 변경하는 것이 필요하다. 본 논문은 무선 채널 특성의 이론적 분석과 저 출력 라디오파를 이용하는 센서(sensor) 네트워크상에서의 실험적 측정을 통해서, 적응적 FEC 기법이 무선 네트워크의 전송 성능을 향상시킬 수 있다는 것을 보여준다. 또한 전송 에러률이 <TEX>$10^{-3}$</TEX> 정도로 상당히 오류가 많은 무선 네트워크에서 적용할 수 있는 적응적 FEC 기법, FECA(FEC-level Adaption)기법을 제안한다. 마지막으로 FEC 알고리즘의 성능 비교를 위해 패킷 시뮬레이션에서 비트 레벨(bit-level) 오류를 시뮬레이션할 수 있는 통합 시뮬레이션 기법을 소개한다. 통합 시뮬레이션을 통해 FECA는 정적 FEC에 비해서 지속적으로 무선 채널 오류률에 알맞는 FEC 정정 코드를 채택하여 성능을 향상시키는 것을 확인하였다. Wireless mobile networks tend to drop a large portion of packets due to propagation errors rather than congestion. To Improve reliability over noisy wireless channels, wireless networks can employ forward error correction (FEC) techniques. Static FEC algorithms, however, can degrade the performance by poorly matching their overhead to the degree of the underlying channel error, especially when the channel path loss rate fluctuates widely. This paper investigates the benefits of an adaptable FEC mechanism for wireless networks with severe packet loss by analytical analysis or measurements over a real wireless network called sensor network. We show that our adaptive FEC named FECA (FEC-level Adaptation) technique improves the performance by dynamically tuning FEC strength to the current amount of wireless channel loss. We quantify these benefits through a hybrid simulation integrating packet-level simulation with bit-level details and validate that FECA keeps selecting the appropriate FEC-level for a constantly changing wireless channel.

Adaptive rate control and QoS provisioning in direct broadcast satellite networks

Adaptive rate control, if properly employed, is an effective mechanism to sustain acceptable levels of Quality of Service (QoS) in wireless networks where channel and traffic conditions vary over time. In this paper we present an adaptive rate (source and channel) control mechanism, developed as part of an Adaptive Resource Allocation and Management (ARAM) algorithm, for use in Direct Broadcast Satellite (DBS) networks. The algorithm performs admission control and dynamically adjusts traffic source rate and Forward Error Correction (FEC) rate in a co-ordinated fashion to satisfy QoS requirements. To analyze its performance, we have simulated the adaptive algorithm with varying traffic flows and channel conditions. The traffic flow is based on a variable bit rate (VBR) source model that represents Motion Picture Expert Group (MPEG) traffic fluctuations while the DBS channel model is based on a two-state Additive White Gaussian Noise (AWGN) channel. For measures of performance, the simulator quantifies throughput, frame loss due to congestion during transmission as well as QoS variations due to channel (FEC) and source (MPEG compression and data transmission) rate changes. To show the advantage of the adaptive FEC mechanism, we also present the performance results when fixed FEC rates are employed. The results indicate significant throughput and/or quality gains are possible when the FEC/source pairs are adjusted properly in co-ordination with source rate changes.

Simulation study of prioritized forward error correction in ATM networks

The dominant source of error in ATM networks is due to buffer overflow in congested conditions, which results in cell loss. To meet the end users' quality of service (QOS) requirements, automatic repeat request (ARQ) and forward error correction (FEC) techniques are two candidates for loss compensation. The FEC technique is appropriate for real-time traffic, although it also causes increases in network loads and loss rates. The situation is worse for non-real-time and loss-sensitive bursty data traffic that experience more congestion when FEC is not applied. In this paper we present a prioritized FEC mechanism combined with selective cell discarding, in which low-priority FEC cells are discarded first in the event of congestion in order to improve the block loss performance. The mechanism is simple to implement, and the block loss rate is reduced compared with the original FEC method.

Transport and display mechanisms for multimedia conferencing across packet-switched networks

Abstract A transport protocol that supports real-time communication of audio/video frames across campus-area packet switched networks is presented. It is a “best effort” protocol that attempts to ameliorate the effect of jitter, load variation, and packet loss, to provide low latency, synchronized audio and video communications. This goal is realized through four transport and display mechanisms, and a real-time implementation of these mechanisms that integrates operating system services (e.g., scheduling and resource allocation, and device management) with network communication services (e.g., transport protocols), and with application code (e.g., display routines). The four mechanisms are: a facility for varying the synchronization between audio and video to achieve continuous audio in the face of jitter, a network congestion monitoring mechanism that is used to control audio/video latency, a queuing mechanism at the sender that is used to maximize frame throughput without unnecessarily increasing latency, and a forward error correction mechanism for transmitting audio frames multiple times to ameliorate the effects of packet loss in the network. The effectiveness of these techniques is demonstrated by measuring the performance of the protocol when transmitting audio and video across congested networks.