Real-Time Media Flow Protocol Research Articles

Peer-Assisted Video Streaming With RTMFP Flash Player: A Measurement Study on PPTV

Real-time media flow protocol (RTMFP) is a protocol developed by Adobe for multimedia delivery under both client–server and peer-to-peer (P2P) paradigms. Currently, major Internet video service providers, such as PPTV and iQIYI, have already built their Web-based video streaming systems with RTMFP. In such a system, a user only needs to install a Flash Player plug-in on his Web browser, and can stream videos in a peer-assisted way. Despite its wide usage, RTMFP has received little attention from the measurement community. In this paper, we select PPTV as an example and study the RTMFP video streaming technology with a measurement approach. We reveal the architecture of PPTV’s RTMFP streaming system and show that, compared with proprietary P2P networks, the RTMFP network has a different content distribution policy, and exhibits different features on peers’ streaming behaviors, potential system bottleneck, and network dynamics. We also study RTMFP’s video transmission and find that the protocol’s selective retransmission scheme can effectively overcome packet losses and improve the video playback quality; however, the TCP-like congestion control mechanism of RTMFP does not lead to fairness between RTMFP and Transmission Control Protocol (TCP) traffics, due to the mismatch between the inherited pull-based video segment distribution model of the P2P streaming application and the protocol’s built-in congestion control mechanism. This paper provides insights into the RTMFP-based video streaming technology and is helpful for people to construct better peer-assisted video systems with RTMFP.

A Novel UDT‐Based Transfer Speed‐Up Protocol for Fog Computing

Fog computing is a distributed computing model as the middle layer between the cloud data center and the IoT device/sensor. It provides computing, network, and storage devices so that cloud based services can be closer to IOT devices and sensors. Cloud computing requires a lot of bandwidth, and the bandwidth of the wireless network is limited. In contrast, the amount of bandwidth required for “fog computing” is much less. In this paper, we improved a new protocol Peer Assistant UDT‐Based Data Transfer Protocol (PaUDT), applied to Iot‐Cloud computing. Furthermore, we compared the efficiency of the congestion control algorithm of UDT with the Adobe’s Secure Real‐Time Media Flow Protocol (RTMFP), based on UDP completely at the transport layer. At last, we built an evaluation model of UDT in RTT and bit error ratio which describes the performance. The theoretical analysis and experiment result have shown that UDT has good performance in IoT‐Cloud computing.

A testbed for experimental evaluation of efficient multimedia delivery over lossy networks

Multimedia transmission over lossy networks has been rapidly increasing, especially with the advent of different high-speed and cost-effective Internet technologies. Despite these technology advances, however, the quality of video streaming over lossy networks is still a major challenge towards obtaining the best quality of service. In this paper, a new experimental testbed that is based on the real-time media flow protocol (RTMFP) is proposed to enable researchers to investigate and optimise the effect of changing multiple key parameters relevant to the video-streaming quality. The impact of these parameters, which include video resolution, frames per second, and compression rate on the video quality are investigated with various settings of network packet loss. The obtained results and relationships between these parameters have led to development of a new algorithm that is capable of changing the parameters under study, in real time, according to the estimated packet loss towards maximum obtainable quality. Design details and experimental setup of the proposed testbed are presented along with key practical considerations and experimental evaluation results.

Network-status aware quality adaptation algorithm for improving real-time video streaming over the internet

Video streaming over Internet has been gaining momentum and several quality adaptation schemes have been reported to improve quality of the streamed videos. Most of these schemes focus on adjusting the video encoding rate to match certain network conditions. This paper presents a new quality adaptation algorithm for real-time video streaming over Internet. The proposed algorithm is based upon simultaneous adaptation of multiple key parameters such as video frame rate, resolution, and frame quality to achieve the best possible video quality and minimize possibility of service interruption in lossy networks. Furthermore, the current network status and motion of the streamed video sequence have also been taken into account throughout the adaptation process. A video conferencing test-bed which incorporates the Adobe Flash Real-Time Media Flow Protocol (RTMFP) is built and utilized to investigate the effect of various combinations of the video parameters under study on the quality of sample video clips streamed at slow, medium and fast motions. A quality-adaptation algorithm based on experimental investigations is then developed and its performance is assessed using both subjective and objective evaluations. The obtained results and observations demonstrate superior performance of the developed adaptation algorithm as compared to equivalent algorithms with fixed video quality settings at similar test conditions.

PROGRAM VIDEO CALL BERBASIS PEER TO PEER MENGGUNAKAN PROTOKOL RTMFP

The rapid development of information technology-based communication to encourage the development of video conferencing applications to a variety of media platforms. One of the media that is easily accessible is a web application development based RIA (Rich Internet Application). In this study developed a flash-based video conferencing application to utilize the net class group in ActionScript 3.0 and multimedia protocols RTMFP (Real Time Media Flow Protocol). Streaming video delivery technology requires codecs to be able to transmit video and audio. To find out which one is suitable codec in the development of this application, the analysis is done where appropriate codec for video conferencing on the public network access in terms of the quality of the resulting. The conclusion of the implementation and analysis of three types of audio codecs and codec, is that the codec h.264 and h.264 base profile play a video codec that is suitable for public networks because it produces little noise when there is a lot of movement when compared to the Sorenson video codec.

An Analysis and Proof on Self‐Similarity Property of Flash P2P Internet Video Traffic

In order to study the amount of bandwidth resource loss caused by flash P2P technique, its influence on network load and its maintenance cost, the characteristic of Internet traffic of flash P2P is studied. On the basis of analysis of features of Real time media flow protocol (RTMFP), the traffic of flash P2P can be identified from the Internet traffic. By extraction and analysis of the traffic of the three largest online video content providers in China (Youku, Iqyi and Sohu Video) and the calculation of probability distribution of the traffic, it is found that the transmission time of flash P2P traffic and the transmission time interval obeys the heavy-tailed distribution. This accounts for the self-similarity of the flash P2P traffic. To better explain the phenomenon, the long-range dependent model is established based on the current traffic. By the established mathematical model, it is strictly proved that flash P2P traffic has self-similar characteristic.