FEC Performances in Multimedia Streaming

In this paper, the performances of packet-level media-independent FEC schemes are computed in terms of both packet loss ratio and average burst length of multimedia data after error recovery. The set of equations leading to the analytical formulation of both parameters are first given for a renewal error process. Finally, the FEC performances parameters are computed in the case of a Gilbert-model loss process and compared to experimental data

Packet loss characteristics for M/G/1/N queueing systems

In this contribution we investigate higher-order loss characteristics for M/G/1/N queueing systems. We focus on the lengths of the loss and non-loss periods as well as on the number of arrivals during these periods. For the analysis, we extend the Markovian state of the queueing system with the time and number of admitted arrivals since the instant where the last loss occurred. By combining transform and matrix techniques, expressions for the various moments of these loss characteristics are found. The approach also yields expressions for the loss probability and the conditional loss probability. Some numerical examples then illustrate our results

Enhancement of Adaptive Forward Error Correction Mechanism for Video Transmission Over Wireless Local Area Network

Video transmission over the wireless network faces many challenges. The most critical challenge is related to packet loss. To overcome the problem of packet loss, Forward Error Correction is used by adding extra packets known as redundant packet or parity packet. Currently, FEC mechanisms have been adopted together with Automatic Repeat reQuest (ARQ) mechanism to overcome packet losses and avoid network congestion in various wireless network conditions. The number of FEC packets need to be generated effectively because wireless network usually has varying network conditions. In the current Adaptive FEC mechanism, the FEC packets are decided by the average queue length and average packet retransmission times. The Adaptive FEC mechanisms have been proposed to suit the network condition by generating FEC packets adaptively in the wireless network. However, the current Adaptive FEC mechanism has some major drawbacks such as the reduction of recovery performance which injects too many excessive FEC packets into the network. This is not flexible enough to adapt with varying wireless network condition. Therefore, the enhancement of Adaptive FEC mechanism (AFEC) known as Enhanced Adaptive FEC (EnAFEC) has been proposed. The aim is to improve recovery performance on the current Adaptive FEC mechanism by injecting FEC packets dynamically based on varying wireless network conditions. The EnAFEC mechanism is implemented in the simulation environment using Network Simulator 2 (NS-2). Performance evaluations are also carried out. The EnAFEC was tested with the random uniform error model. The results from experiments and performance analyses showed that EnAFEC mechanism outperformed the other Adaptive FEC mechanism in terms of recovery efficiency. Based on the findings, the optimal amount of FEC generated by EnAFEC mechanism can recover high packet loss and produce good video quality

A two-level Markov model for packet loss in UDP/IP-based real-time video applications targeting residential users

The packet loss characteristics of Internet paths that include residential broadband links are not well understood, and there are no good models for their behaviour. This compli- cates the design of real-time video applications targeting home users, since it is difficult to choose appropriate error correction and concealment algorithms without a good model for the types of loss observed. Using measurements of residential broadband networks in the UK and Finland, we show that existing models for packet loss, such as the Gilbert model and simple hidden Markov models, do not effectively model the loss patterns seen in this environment. We present a new two-level Markov model for packet loss that can more accurately describe the characteristics of these links, and quantify the effectiveness of this model. We demonstrate that our new packet loss model allows for improved application design, by using it to model the performance of forward error correction on such links

Impact of large block FEC with different queue sizes of drop tail and RED queue policy on video streaming quality over internet

In this paper, we report an investigation on the impact of large block Forward Error Correction (FEC) with Drop Tail (DT) and Random Early Detection (RED) queue policies on network performance and quality of video streaming.FEC is a technique that uses redundant packets to reconstruct dropped packets, while DT and RED are the most popular queue management policies used in network routers.DT mainly depends on the size of the queue buffer to decide on whether to drop a packet or not.RED monitors the average queue size and drops arriving packets probabilistically.The probability of dropping a packet increases as the estimated average queue size grows.In the investigation, we consider simulation settings with varying size of queue buffers.Results obtained from the simulation experiments show that large block FEC and queue size affect the performance the network.Consequently, the qualities of multimedia applications are also affected

Perceived Quality of Packet Audio under Bursty Losses

We examine the impact of bursty losses on the perceived quality of packet audio, and investigate the effectiveness various approaches to improve the quality. Because the degree of burstiness depends on the packet interval, we first derive a formula to re-compute the conditional loss probability of a Gilbert loss model when the packet interval changes. We find that FEC works better at a larger packet interval under bursty losses. In our MOS-based (Mean Opinion Score) listening tests, we did not find a consistent trend in MOS when burstiness increases if FEC is not used. That is, In some occasions MOS can be higher with a higher burstiness. With FEC, our results confirms the analytical results that quality is better with a larger packet interval, but T should not be too large to avoid severe penalty on a single packet loss. We also find that low bit-rate redundancy generally produces lower perceived quality than FEC, if the main codec is already a low bit-rate codec. Finally, we compare our MOS results with objective quality estimation algorithms (PESQ, PSQM/PSQM+, MNB and EMBSD). We find PESQ has the best linear correlation with MOS, but the value is still less than commonly cited, implying they cannot be used in isolation to predict MOS

Flexible Macroblock Ordering for Context-Aware Ultrasound Video Transmission over Mobile WiMAX

The most recent network technologies are enabling a variety of new applications, thanks to the provision of increased bandwidth and better management of Quality of Service. Nevertheless, telemedical services involving multimedia data are still lagging behind, due to the concern of the end users, that is, clinicians and also patients, about the low quality provided. Indeed, emerging network technologies should be appropriately exploited by designing the transmission strategy focusing on quality provision for end users. Stemming from this principle, we propose here a context-aware transmission strategy for medical video transmission over WiMAX systems. Context, in terms of regions of interest (ROI) in a specific session, is taken into account for the identification of multiple regions of interest, and compression/transmission strategies are tailored to such context information. We present a methodology based on H.264 medical video compression and Flexible Macroblock Ordering (FMO) for ROI identification. Two different unequal error protection methodologies, providing higher protection to the most diagnostically relevant data, are presented

Adaptive and robust media streaming over multiple channels with bursty losses

This paper addresses the problem of efficiently delivering a layered media stream from multiple senders to a single receiver, over channels that present correlated packet loss patterns. Using a digital fountain approach, the performance of a distributed streaming system is driven by the probability of receiving a given number of packets on aggregate over the multiple channels. In addition, such a system allows to avoid the need for communication between streaming servers. We devise an optimization problem whose solution provides the optimal number of packets that need to be transmitted per channel, in order to maximize the probability of correct decoding for a given media stream. Our findings indicate that it is in general important to consider both the Packet Loss Ratio (PLR) and Average Burst Length (ABL) in channel selection problems such as multipath routing or rate aggregation on multiple bursty channels. Finally we present a low-complexity algorithm which is able to quickly find a suboptimal yet effective solution to the combinatorial optimization problem