This paper addresses the problem of streaming packetized media

Abstract—Due to the explosive growth of the Internet and increasing demand for multimedia information on the web, streaming video over the Internet has received tremendous attention from academia and industry. Transmission of real-time video typically has bandwidth, delay, and loss requirements. However, the current best-effort Internet does not offer any quality of service (QoS) guarantees to streaming video. Furthermore, for video multicast, it is difficult to achieve both efficiency and flexibility. Thus, Internet streaming video poses many challenges. To address these challenges, extensive research has been conducted. This special issue is aimed at dissemination of the contributions in the field of streaming video over the Internet. To introduce this special issue with the necessary background and provide an integral view on this field, we cover six key areas of streaming video. Specifically, we cover video compression, application-layer QoS control, continuous media distribution services, streaming servers, media synchronization mechanisms, and protocols for streaming media. For each area, we address the particular issues and review major approaches and mechanisms. We also discuss the tradeoffs of the approaches and point out future research directions. Index Terms—Application-layer QoS control, continuous media distribution services, Internet, protocol, streaming video,

This paper introduces a new framework for video content delivery that is based on the transcoding of multiple video objects. Generally speaking, transcoding can be defined as the manipulation or conversion of data into another more desirable format. In this paper, we consider manipulations of object-based video content, and more specifically, from one set of bit streams to another. Given the object-based framework, we present a set of new algorithms that are responsible for manipulating the original set of video bit streams. Depending on the particular strategy that is adopted, the transcoder attempts to satisfy network conditions or user requirements in various ways. One of the main contributions of this paper is to discuss the degrees of freedom within an object-based transcoder and demonstrate the flexibility that it has in adapting the content. Two approaches are considered: a dynamic programming approach and an approach that is based on available meta-data. Simulations with these two approaches provide insight regarding the bit allocation among objects and illustrates the tradeoffs that can be made in adapting the content. When certain meta-data about the content is available, we show that bit allocation can be significantly improved, key objects can be identified, and varying the temporal resolution of objects can be considered.

Available bandwidth is the most useful measurement to network adaptive applications and transports. Unfortunately, it is also very difficult to measure in a network that cannot be well approximated by a weighted fair queuing model, such as the Internet. At the same time, the Internet scalable architecture calls for end-to-end measurements. Current available bandwidth measurement techniques are based in observing packet dispersion in a packet train or pair. The available bandwidth is sampled by using a "bytes divided by dispersion" (or "bytes over time") calculation and then filtered. In this paper, we deal with sampling the available bandwidth from packet dispersion. We argue that existing techniques, being heuristics, are unintuitive, are not based on any network model, and introduce an error in the process right from the sampling. We propose a different calculation of the available bandwidth from packet dispersion observation, we call ab-probe. It is also simple and as robust, and follows a simple network model. It is an improvement to the "bytes over time" calculation because it also captures and deducts a portion of the argued sample calculation error. We study this new model by exploring the differences between the two methods and its robustness to parameter estimation errors. We then apply our model to a well known available bandwidth measurement technique called cprobe. We find that our model explains cprobe's behavior as recorded by its researchers' published real experiments. We further validate the measurement using simulations in some simple wired scenarios.

Abstract. Available bandwidth knowledge is very useful to network protocols. Unfortunately, available bandwidth is also very difficult to measure in packet networks, where methods to guarantee and keep track of the bandwidth (eg, weighted fair queuing scheduling) do not work well, for example the Internet. In this paper we are dealing with an available bandwidth sampling technique based on the observation of packet time dispersion in a packet train or pair. In standard techniques the available bandwidth is sampled by using a “bytes divided by dispersion ” (or “bytes over time”, BoT) calculation and then filtered. This method of calculating available bandwidth samples has been used in all packet dispersion related work. We propose a new sampling method of available bandwidth called ab-probe. The ab-probe method uses an intuitive model that helps understand and correct the error introduced by the BoT sample calculation. We theoretically compare the new model with the previous one, exploring their differences, observability and robustness. We argue that the model may significantly improve protocols that can use an available bandwidth measurement, in particular transport-level protocols that currently use the BoT calculation. 1

Abstract • Multimedia transmission over wireless Internet is a challenging work since the random errors frequently occurred in the wireless links heavily deteriorates the performance of traditional transport protocols such as TCP and UDP. This paper presents a new scheme for delivering multimedia over a connection of which only the last hop is wireless link. Simultaneously error control and congestion control are addressed in this wireless Internet platform. The packet loss caused by Internet congestion and wireless error can be differentiated in our scheme. Fading and high bit error rate (BER) on the wireless link as well as congestion in the wired network are separately controlled. Simulation results demonstrate that the proper congestion control and error control can improve throughput of the connection and the end-to-end quality of delivered media. 1.

A content-aware rate controller (CRC) for MPEG-4 FGS stored video streaming over the Internet is proposed. The CRC optimally uses available bandwidth and client buffer to produce smooth video quality and provide protection to video segments with important content. The FGS video stream is forward-shifted to client buffer by actively dropping high enhancement layers. When bandwidth decreases sharply, the FS buffer is used as bandwidth so that the sharp bandwidth drop is hidden from the decoder. The active dropping and forward-shifting process is restarted when bandwidth recovers. Thus, higher layers of less importance content are dropped and lower layers of more importance are protected. A priori information about the stream content is used in the CRC such that video segments with important content are protected by allocating them more bandwidth.

OF THE DISSERTATION Adaptive Multimedia in Wireless IP Networks by Matheos Ioannis Kazantzidis Doctor of Philosophy in Computer Science University of California, Los Angeles, 2002 Professor Mario Gerla, Chair Support for video and audio applications is important to single and multi hop wireless networks whether they are used as extensions to the Internet or not. Due to the variability of response of the air medium and the mobility support that is expected of these networks, it is accepted that such applications must dynamically adapt to network conditions, taking advantage of the different content representations achieved by advances in coding. This adaptability targets at maximizing the overall QoS delivered by the network and may be classified into (i) The transport functionality that decides the network parameters e.g. sending rate and (ii) The presentation functionality that decides the content that should fit the network parameters. In wireless, it is particularly difficult to implement an accurate monitoring process (measurement) and embed it into a xv distributed strategy that efficiently controls the scarce network resources. Therefore, transport protocols designed for wired networks fail. This, combined with scalability challenges of some ad hoc environments (e.g. battlefield) motivates the exploration of an important trade-off for this environment. On the one hand, adopting a thin and scalable network architecture allows for end-to-end adaptation which is limited in measurement accuracy and consequently performance. On the other hand, the implementation of lower layer feedback support leads to architectures that are less scalable and bear a higher deployment cost. But, can deal effectively with the measurement inaccuracy problem. In order to explore th...

Layered multicast is an efficient technique to deliver video to users over wired and wireless networks.

In this paper, we examine the performance of specific implementations of multiple description coding and of layered coding for video streaming over error-prone packet switched networks. We compare their performance using different transmission schemes with and without network path diversity. It is shown that, given specific implementations, there is a large variation in relative performance between multiple description coding and layered coding depending on the employed transmission scheme. For scenarios where the packet transmission schedules can be optimized in a rate-distortion sense, layered coding provides a better performance. The converse is true for scenarios where the packet schedules are not rate-distortion optimized.