Dynamic Quality Adaptation Mechanisms for TCP-friendly MPEG-4 Video Transfer (original) (raw)
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IP Video Streaming with Fine-Grained TCP-Friendly Rate Adaptation
Lecture Notes in Computer Science, 2003
Video streaming over the Internet is a challenging task since the Internet is a shared environment offering only best effort service. That is, it offers no quality of service and no guarantee of resources in term of (1) bandwidth, (2) transfer delay, (3) delay variation (jitter), and (4) packet losses. Then, network stability and traffic fairness become critical issues. To solve these problems some source rate control and adaptation should be introduced for UDP traffic as well, in such a way that this traffic becomes TCP-compatible "TCP-friendly". In this article we propose an adaptive streaming framework for unicast MPEG-4 streams over TCP/IP networks. Based on Audio-Visual Content (AVOs) classification and network congestion feedback, video sources dynamically adds and drops MPEG-4 AVO to the streamed multiplex to conform to the TCP-Friendly Rate Control (TFRC) mechanism. Using a content classification model, TFRC automatically adjusts the number of AVOs to be streamed to adapt to network congestion while given much attention to the quality of the service perceived by the end-user. To achieve such a dynamic output rate and video quality adjustment, MPEG-4 AVOs are classified and multiplexed according to both application-level QoS parameters and AVOs semantic descriptors. AVOs requiring same QoS from the network are automatically classified and mapped to one of the available IP DiffServ PHB (Per Hop Behaviors). Performance evaluation shows that transmitted video gracefully adapts to network bandwidth variations while optimizing user perceived quality.
MPEG-TFRCP: Video transfer with TCP-friendly rate control protocol
2001
As the use of real-time multimedia applications increases, bandwidth available to TCP connections is oppressed by "greedy" UDP traffic and their performance extremely deteriorates. In order that both TCP and UDP sessions fairly co-exist in the Internet, UDP sessions should properly react against congestion as TCP. In this work, we implement a "TCP-friendly" rate control mechanism suitable to video applications and investigate its applicability to a real system through observation of the video quality at the receiver. It is shown through our experimental system that we can achieve high-quality and stable video transfer while fairly sharing the network bandwidth with TCP by applying our rate control at a control interval of 16 or 32 times as long as RTT.
Internet Quality and Performance and Control of Network Systems, 2001
When both TCP and UDP connections co-exist in the Internet environment, the performance of TCP connections is heavily affected by the behavior of "greedy" UDP connections of real-time multimedia applications. In this paper, we propose a new TCP-friendly rate control protocol for video connections, called MPEG-TFRCP, to fairly share the link with TCP connections. To achieve fairness among TCP and UDP connections while performing high quality video transmission, we argue that (1) the interval of rate control must be appropriately determined, (2) the network condition must be accurately predicted, (3) the TCP throughput must be precisely estimated and (4) the video rate must be effectively adjusted. Although our algorithm is based on the existing proposals which do not satisfy all of those conditions, through careful considerations on the applicability of TFRCP to the actual video applications ours can achieve the high-quality MPEG-2 video transfer while satisfying the TCP-friendliness.
Packet loss resilient MPEG-4 compliant video coding for the Internet
Signal Processing: Image Communication, 1999
Targeting multimedia communications over the Internet, this paper describes a set of complementary techniques in the direction of both improved packet loss resiliency of video-compressed streams and e$cient usage of available network resources. Aiming "rst at a best trade-o! between compression e$ciency and packet loss resiliency, a procedure for adapting the video coding modes to varying network characteristics is introduced. The coding mode selection is based on a rate-distortion procedure with global distortion metrics incorporating channel characteristics under the form of a two-state Markov model. This procedure has been incorporated in an MPEG-4 video encoder. It has been observed that, in error-free environments, the channel adaptive mode selection technique does not bring any penalty in terms of compression, with respect to the initial MPEG-4 encoder, while allowing a signi"cant gain with respect to simple conditional replenishment. On the other hand, under the same loss conditions, it is shown that this procedure signi"cantly improves the encoder's performance with respect to the original MPEG-4 encoder, to approach the robustness of conditional replenishment mechanisms. This intrinsic robusti"cation of the encoder allows to minimize the e!ects of packet losses on the visual quality of the received video; however, it does not avoid losses. A rate-based #ow control mechanism is then developed and introduced into the encoder, in order to match the bandwidth requirements of the source to the bandwidth available over the path of the connection, for both &social' and &individual' bene"ts. The control mechanism developed combines an RTT-based control loop allowing early reaction to congestion and a TCPfriendly rate prediction model getting into play under lossy conditions. This hybrid control mechanism allows full rate control (even in loss-free conditions) and smooth rate variations together with high responsiveness. The introduction of the rate control in the MPEG-4 compliant encoder allows to maintain a stable PSNR and visual quality while decreasing signi"cantly the source throughput, hence reducing congestion and loss provoked by the same video source at a constant bit-rate. (F. Le LeH annec) with respect to quality of service, congestion management, and network friendliness. Due to the real-time nature of envisioned data streams, multimedia delivery usually makes use of the so-called unresponsive transport protocols, i.e. the User Datagram Protocol (UDP) and/or Real-time Transport Protocol (RTP). Both UDP and RTP o!er no quality of service control mechanisms and can therefore not guarantee any level of QoS, 0923-5965/99/$ -see front matter 1999 Elsevier Science B.V. All rights reserved. PII: S 0 9 2 3 -5 9 6 5 ( 9 9 ) 0 0 0 2 3 -5
Quality-Driven TCP Friendly Rate Control for Real-Time Video Streaming
Global Telecommunications Conference, . GLOBECOM . IEEE, 2008
TCP Friendly Rate Control (TFRC) has been designed to provide smoother sending rate than TCP for multimedia applications. However, most existing work on TFRC is restricted within exploring the performance of TFRC itself in wired or wireless networks without considering the interaction between TFRC and other network layers. This paper proposes a quality- driven TFRC framework for real-time video streaming,
Enhanced TCP-Friendly Rate Control for Supporting Video Traffic over Internet
Video traffic nowadays forms the majority of traffic over the Internet, and is predicted to be the most prevailing traffic type in the coming few years. TCP Friendly Rate Control (TFRC) is one of the most promising end-to-end congestion control protocols that are intended for unicast playback of Internet streaming applications. This paper presents a new TCP-Friendly congestion control protocol, called Enhanced TCP-Friendly Rate Control (ETFRC), for supporting real time video traffi c over the Internet. The proposed protocol is developed by adjusting the sending rate, at the sender side, dynamically based on the current state of the network, and the current state of the receiver. In other words, ETFRC embodies a new algorithm to tune (increase or decrease) the sending rate, at the sender side, according to the difference between the calculated rate by the sender and the reported rate from the receiver side. The performance of the proposed ETFRC protocol is evaluated using the network simulator ns-2 considering different scenarios. In these scenarios, simulated video traffic from the Evalvid framework is sent over the designed topology and different performance parameters are measured and compared with that obtained by applying the original TFRC protocol. The simulation results show that ETFRC performance surpassed TFRC in terms of throughput, jitter, and packet loss.
Quality Adaptation For MPEG4 FGS Video Streaming Over Internet
2000
A quality adaptation mechanism for MPEG-4 FGS video streaming is proposed. It decides the way FGS video is truncated at the server. The client receives the truncated video stream and decodes it. In any streaming application the system has to ensure that the streaming proceeds in a time synchronized maner, the quality adaptation algorithm tries to handle this misalignment and
2000
When both TCP and UDP connections co-exist in the Internet environment, the performance of TCP connections is heavily affected by the behavior of "greedy" UDP connections of real-time multimedia applications. In this paper, we propose a new TCP-friendly rate control protocol for video connections, called MPEG-TFRCP, to fairly share the link with TCP connections. To achieve fairness among TCP and
Adaptive quality of service for streamed MPEG-4 over the Internet
ICC 2001. IEEE International Conference on Communications. Conference Record (Cat. No.01CH37240), 2001
In streaming technologies, multimedia data is continuously fed to the user while they are processing (viewing, listening to, reading) the data. However, streaming with a specified Quality of Service (QoS) is not yet a solved problem. We are building a prototype system to adapt multimedia streaming to a fluctuating network load and/or client requests, thereby providing adaptive QoS. We have shown the feasibility of adjusting QoS levels during transmission by dynamically modifying which transmission layers are actually sent, as well as the transmission profiles and packet priority ratios.