Kazuhisa Matsuzono

Performance analysis of a high-performance real-time application with several AL-FEC schemes

Real-time streaming applications typically require minimizing packet loss and transmission delay so as to keep the best possible playback quality. From this point of view, IP datagram losses (e.g. caused by a congested router, or caused by a short term fading problem with wireless transmissions) have major negative impacts. Although Application Layer Forward Error Correction (AL-FEC) is a useful technique for protecting against packet loss, the playback quality is largely sensitive to the AL-FEC code/codec features and the way they are used. In this work, we consider three FEC schemes for the erasure channel: 2D parity check codes, Reed-Solomon over GF(28) codes, and LDPC-Staircase codes, all of them being currently standardized within IETF. We have integrated these FEC schemes in the FECFRAME framework, a framework that is also being standardized at IETF, and whose goal is to integrate AL-FEC schemes in real-time protocol stacks in a simple and flexible way. Then we modified the Digital Video Transport System (DVTS) high-performance real-time video streaming application so that it can benefit from FECFRAME in order to recover from transmission impairments. We then carried out several performance evaluations in order to identify, for a given loss rate, the optimal configuration in which DVTS performs the best.

Adaptive Rate Control with Dynamic FEC for Real-Time DV Streaming

For higher streaming quality, a data sender adjusts the data transmission rate according to the network condition between the sender and receiver. The sender and the receiver exchange information about the network condition to decide the appropriate transmission rate. However, monitoring each data flow in real time is difficult, and controlling the sender to adjust the best quality for each receiver in a heterogeneous environment is a real challenge. In this paper, we study packet loss patterns and FEC recovery rate upon data transmission in a congested network, and define an adaptive rate control mechanism that dynamically adjusts the transmission rate and the FEC encoding rate. We then show our adaptive DV Transmission System (DVTS) that supports appropriate rate control for provisioning multimedia streaming with the best possible quality, and evaluate the system on top of our testbed network.

DP-FEC: Dynamic Probing FEC for High-Performance Real-Time Interactive Video Streaming

High-quality and high-performance real-time interactive video streaming requires both keeping the highest data transmission rate and minimizing data packet loss to achieve the best possible streaming quality. TCP-friendly rate control (TFRC) is the most widely recognized mechanism for achieving relatively smooth data transmission while competing fairly with TCP flows. However, because its data transmission rate depends largely on packet loss conditions, high-quality real-time streaming suffers from a significant degradation of streaming quality due to both a reduction in the data transmission rate and data packet losses. This paper proposes the dynamic probing forward error correction (DP-FEC) mechanism that is effective for high-quality real-time streaming to maximize the streaming quality in a situation in which competing TCP flows pose packet losses to the streaming flow. DP-FEC estimates the network condition by dynamically adjusting the degree of FEC redundancy while trying to recover lost data packets. It effectively utilizes network resources and adjusts the degree of FEC redundancy to improve the playback quality at the user side while minimizing the performance impact of competing TCP flows. We describe the DP-FEC algorithm and evaluate its effectiveness using an NS-2 simulator. The results show that by effectively utilizing network resources, DP-FEC enables to retain higher streaming quality while minimizing the adverse condition on TCP performance, thus achieving TCP friendliness.

Analysis of FEC Function for Real-Time DV Streaming

Due to the dissemination of high speed DSL and FTTH, real-time streaming applications transmitting high quality audio and video data have been commonly used on the Internet. These applications usually have strict requirements on delay and packet loss. However, it is difficult to fully satisfy the requirements in general, since the end users cannot guarantee the streaming quality on the best-effort Internet. In this paper, we investigate an FEC (Forward Error Correction) function on DV streaming. We study the relation between network bandwidth and FEC recovery rate upon data transmission, and the receivers play quality and FEC calculation cost. Our experimental results given on top of our testbed network show that the FEC function can provide the best possible streaming quality, without leading the further disruption of video and audio irrespective of the available network bandwidth.

GENEVA: Streaming Control Algorithm Using Generalized Multiplicative-increase/additive-decrease

Motivated by the deployment of wide-area high-speed networks, we propose GENEVA, the streaming control algorithm using generalized multiplicative-increase/additive-decrease (GMIAD). Because current typical con- gestion controllers such as a TCP-friendly rate control prevent occurrences of network congestion reacting susceptibly to packet loss, it causes a significant degradation of streaming quality due to low-achieving throughput (i.e., lower throughput than the maximum throughput that a streaming flow requires in maximum audio/video quality) and data packet losses. GENEVA avoids this problem by allowing a streaming flow to maintain moderate network congestion while trying to recover lost data packets that other competing flows cause during the process of probing for available bandwidth. Using the GMIAD mechanism, the FEC window size (the degree of FEC redundancy per unit time) is adjusted to suppress occurrences of bursty packet loss, while trying to effectively utilize network resources that other competing flows cannot consume due to reductions in the transmission rate in response to packet loss. We describe the GENEVA algorithm and evaluate its effectiveness using an NS-2 simulator. The results show that GENEVA enables high-performance streaming flows to retain higher streaming quality under stable conditions while minimizing the adverse impact on competing TCP performance.