We present ageneral anal[A/v4 model for predicting the reconstructed frame rate of an MPEG stream. Our model captures the temporal relJx#v4[7A# between I-, P, and Bframes but is independent of the channel and media characteristics. We derive an adaptive FEC scheme from the general model and verify it by comparing it to the resul/ of a simulU7A]v The prediction error of the model compared to the simulx'A/ for a wide array of parametervalet isl#U than 5%. We then use the derived adaptive FEC scheme to study the optimal rate ale cation (i.e., between generating a higher frame rate or increasing the protection for a l wer frame rate) when equation-based TCP rate control is used tocoupl packet rates to channel characteristics such as round trip time and packetl ossprobabil4'['J Surprisingl , we find that optimal protectionlrote for I- and P-frames are relx#x/v4 static aslvJ rates increase from 1% to 4%whil changes in the frame type pattern are used to amel/#xv4 the effects of the increasedl oss. The study demonstrates how our model can be used to reveal joint source/channel coding tradeoffs and how theyrelvJ to encoding and transmission parameters.

The growth in power and connectivity of today’s PCs promises a continued increase in the growth of streaming media over the Internet. Hand-in-hand with the increase in streaming media comes the impending threat of unresponsive UDP traffic, often cited as the major threat to the stability of the Internet. The responsiveness of commercial streaming media applications, such as RealNetworks’ RealPlayer, will play an important role in the network impact of streaming media. Unfortunately, there are few empirical studies that analyze the responsiveness, or lack of it, of commercial streaming media. In this work, we measure the responsiveness of RealVideo over UDP by measuring the performance of numerous streaming video clips selected from a variety of RealServers on the Internet. By varying the bottleneck bandwidth to the player, we are able to analyze the TCP-Friendliness of RealVideo over UDP and correlate the results with network and application layer statistics. We find that most streaming RealVideo clips are not bandwidth constrained for typical broadband connections. In times of congestion, most RealVideo UDP streams respond to Internet congestion by reducing the application layer encoding rate, and streams with a minimum encoding rate less than the fair bandwidth share usually achieve a TCP-Friendly rate. In addition, our analysis suggests that a reason streaming applications choose not to use TCP is that the TCP API hides network information, such as loss rate and round-trip time, making it difficult to estimate the available bandwidth for effective media scaling.

End-to-end QoS control over best-effort and differentiated service networks which exhibit variability in their exported service properties looms as an important challenge. In previous work, we have shown how packet-level adaptive FEC can be used in dynamic networks to facilitate invariant user-specified QoS in an end-to-end manner. This paper addresses two important problems---self-similar burstiness and performance degradation of reactive controls subject to long feedback loops---complementing the stability/optimality considerations studied earlier. First, for adaptive redundancy control to be effective, its susceptibility to correlated packet drops and queueing delays stemming from selfsimilar burstiness must be fortified. Second, to preserve FEC's viability over ARQ when transporting real-time traffic in WANs, proactivity must be injected to offset the performance degradation of reactive feedback controls when subject to long RTTs. In this paper, we use the recently advanced multi...

In this paper we present a QoS-Directed Error Control scheme (QDEC) for video multicast in wireless networks. In order to provide multimedia services to mobile users, it is necessary to have not only sufficient bandwidth, but also effective control over wireless channel errors, which are characterized as bursty and location-dependent. The QDEC scheme includes (1) a video QoS differentiation specification to direct the error control

The growth in power and connectivity of today 's PCs promises a continued increase in streaming media over the Internet. Hand-in-hand with the increase in streaming media comes the impending threat of unresponsive traffic, often cited as the major threat to the stability of the Internet. The responsiveness of commercial streaming media applications will play an important role in the network impact of streaming media. Unfortunately, there are few empirical studies that analyze the responsiveness, or lack of it, of current streaming media products. In this work, we measure the responsiveness of RealVideo over UDP compared with RealVideo over TCP by simultaneously playing video clips selected from numerous RealServers on the Internet to two distinct video clients along the same network path. By varying the bottleneck bandwidth to the clients, we are able to analyze the "head-to-head" performance of RealVideo over UDP as compared to RealVideo over TCP, and correlate the results with network and application layer statistics. We find that most streaming RealVideo clips are not bandwidth constrained for typical broadband connections, resulting in a fair share of link bandwidth used by both RealVideo over TCP and RealVideo over UDP. In times of congestion, most RealVideo over UDP does respond to Internet congestion by reducing the application layer encoding rate, often achieving a TCP-Friendly rate. In times of severe congestion, RealVideo over UDP gets a proportionately larger share of available bandwidth than does the same video over TCP.

Measurements of network tra c have shown that self-similarity is a ubiquitous phenomenon spanning across diverse network environments. In previous work, we have explored the feasibility of exploiting long-range correlation structure in self-similar tra c for congestion control. We have advanced the framework of multiple time scale congestion control and shown its e ectiveness at enhancing performance for rate-based feedback control. In this paper, we extend the multiple time scale control framework to window-based congestion control, in particular, TCP. This is performed by interfacing TCP with a large time scale control module which adjusts the aggressiveness of bandwidth consumption behavior exhibited by TCP as a function of \large time scale " network state, i.e., information that exceeds the horizon of the feedback loop as determined by RTT. How to e ectively utilize such information|due to its probabilistic nature, dispersion over multiple time scales, and a ection on top of existing window-based congestion controls|is a nontrivial problem. Our contribution is threefold. First, we de ne a modular extension of TCP|a function call with a simple interface|that applies to various avours of TCP|e.g., Tahoe, Reno, Vegas|

Abstract — Error correction mechanisms enable control and other real-time applications to be executed over unreliable packet-switched networks. By adding carefully adjusted redundancy to transmitted data at the sender, it is possible to recover lost data at the receiver and thereby improve effective throughput. We describe simple models for packet loss, which allow us to find the optimal redundancy as a function of packet loss probability. Two feedforward control mechanisms based on the packet loss probability are presented: one that is computed off-line and another one using an on-line algorithm. A drawback with these approaches is their dependency on accurate network state information and precise loss models. To cope with these issues, we propose a new feedback solution that tracks the optimum using gradient estimation. Simulations in ns-2 illustrate the behavior of the error correction schemes, demonstrating that the feedback solution outperforms the feedforward solution in presence of model errors. I.

New TCP-Friendly constraints require multimedia flows to reduce their data rates under packet loss to that of a conformant TCP flow. To reduce data rates while preserving real-time playout, temporal scaling can be used to discard the encoded multimedia frames that have the least impact on perceived video quality. To limit the impact of lost packets, Forward Error Correction (FEC) can be used to repair frames damaged by packet loss. However, adding FEC requires further reduction of multimedia data, making the decision of how much FEC to use of critical importance. Current approaches use either inflexible FEC patterns or adapt to packet loss on the network without regard to TCP-Friendly data rate constraints. In this paper, we analytically model the playable frame rate of a TCP-Friendly MPEG stream with FEC and temporal scaling, capturing the impact of distributing FEC within MPEG frame types with interframe dependencies. For a given network condition and MPEG video encoding, we use our model to exhaustively search for the optimal combination of FEC and temporal scaling that yields the highest playable frame rate within TCP-Friendly constraints. Analytic experiments over a range of network and application conditions indicate that adjustable FEC with temporal scaling can provide a significant performance improvement over current approaches. Extensive simulation experiments based on Internet traces show that our model can be effective as part of a streaming protocol that chooses FEC and temporal scaling patterns that meet dynamically changing application and network conditions.

We address a new error-resilient scheme for broadcast-quality interactive MPEG-2 video streams to be transmitted over lossy packet networks. A new scene-complexity adaptive mechanism, namely Adaptive MPEG-2 Information Structuring and Protection (AMISP) is introduced. AMISP lies on an information structuring scheme which modulates the number of resynchronization points (i.e., slice headers and intra-coded macroblocks) in order to maximize the perceived video quality. The video quality the end-user experiences depends both on the quality of the compressed video before transmission and on the degradation due to packet loss. Therefore, the structuring scheme constantly determines the best compromise between the rate allocated to encoding pure video information and the rate aiming at reducing the sensitivity to packet loss. It is then extended with a Forward Error Correction (FEC) based protection algorithm to become AMISP. AMISP triggers the insertion of FEC packets in the MPEG-2 video pa...

The Internet, traditionally FTP, e-mail and Web traffic, is increasingly supporting emerg-ing applications such as IP telephony, video conferencing and online games. These new genres of applications have different requirements in terms of throughput and delay than traditional applications. For example, interactive multimedia applications, unlike tra-ditional applications, have more stringent delay constraints and less stringent loss con-straints. Unfortunately, the current Internet offers a monolithic best-effort service to all applications without considering their specific requirements. Adaptive RED (ARED) is an Active Queue Management (AQM) technique, which optimizes the router for through-put. Throughput optimization provides acceptable QoS for traditional throughput sensi-tive applications, but is unfair for these new delay sensitive applications. While previous work has used different classes of QoS at the router to accommodate applications with varying requirements, thus far all have provided just 2 or 3 classes of service for ap-plications to choose from. We propose two AQM mechanisms to optimize router for better overall QoS. Our first mechanism, RED-Worcester, is a simple extension to ARED