This paper presents pathChirp, a new active probing tool for estimating the available bandwidth on a communication network path. Based on the concept of "self-induced congestion," pathChirp features an exponential flight pattern of probes we call a chirp. Packet chips offer several significant advantages over current probing schemes based on packet pairs or packet trains. By rapidly increasing the probing rate within each chirp, pathChirp obtains a rich set of information from which to dynamically estimate the available bandwidth. Since it uses only packet interarrival times for estimation, pathChirp does not require synchronous nor highly stable clocks at the sender and receiver. We test pathChirp with simulations and Internet experiments and find that it provides good estimates of the available bandwidth while using only a fraction of the number of probe bytes that current stateof -the-art techniques use.

Video transmission in wireless environments is a challenging task calling for high-compression efficiency as well as a network friendly design. Both have been major goals of the H.264/AVC standardization effort addressing "conversational" (i.e., video telephony) and "nonconversational" (i.e., storage, broadcast, or streaming) applications. The video compression performance of the H.264/AVC video coding layer typically provides a significant improvement. The network-friendly design goal of H.264/AVC is addressed via the network abstraction layer that has been developed to transport the coded video data over any existing and future networks including wireless systems. The main objective of this paper is to provide an overview over the tools which are likely to be used in wireless environments and discusses the most challenging application, wireless conversational services in greater detail. Appropriate justifications for the application of different tools based on experimental results are presented.

The loss and delay experienced by packets travelling along an Internet network path are mainly governed by the characteristics of a bottleneck link, such as available data rate and queue size. In this work, we propose a framework for rate-distortion optimized packet scheduling with adaptive rate control for media streaming over bandwidth-constrained bottleneck links. The framework computes optimal packet schedules while continuously adapting its instantaneous rate to the following three factors: the available data rate and the current queue size on the bottleneck link, and the congestion that packets transmitted under the schedules will create on the bottleneck link. Experimental results demonstrate that our framework does not lose in rate-distortion performance over rate-distortion optimized packet scheduling without strict rate control, while producing at the same time a much smoother instantaneous rate feeding the bottleneck queue. This in turn contributes to fairness to other flows sharing the bottleneck link and causes less variations in queue size, thereby avoiding queue overflow and unnecessarily long packet delays on the bottleneck link. 1.

Video communication is often afflicted by various forms of losses, such as packet loss over the Internet. This paper examines the question of whether the packet loss pattern, and in particular the burst length, is important for accurately estimating the expected mean-squared error distortion. Specifically, we (1) verify that the loss pattern does have a significant effect on the resulting distortion, (2) explain why a loss pattern, for example a burst loss, generally produces a larger distortion than an equal number of isolated losses, and (3) propose a model that accurately estimates the expected distortion by explicitly accounting for the loss pattern, inter-frame error propagation, and the correlation between error frames. The accuracy of the proposed model is validated with JVT/H. 26L coded video and previous frame concealment, where for most sequences the total distortion is predicted to within -4-0.25 dB for burst loss of length two packets, as compared to prior models which underestimate the distortion by about 1.5 dB. Furthermore, as the burst length increases, our prediction is within -4-0.7 dB, while prior models degrade and underestimate the distortion by over 3 dB.

Scalable, or layered, media representation appears to be more suitable for transmission over the current heterogeneous networks. In this paper we study the problem of scalable layered streaming media delivery over a lossy channel. The goal is to find an optimal transmission policy to achieve the best playback quality at the client end. The problem involves some trade-offs such as time-constrained delivery and data dependencies. For example, a layer should be dropped before transmission if it already has a delay such that it cannot be played before its scheduled time. Moreover, less important layers with near-playback-time may also be dropped or delayed for delivery in order to save bandwidth for other layers with a high priority. We propose a framework for scalable streaming media delivery, that involves a novel scheduling algorithm called Expected runtime Distortion Based Scheduling, EDBS, which decides the order in which packets should be transmitted in order to improve client playback quality in the presence of channel losses. A fast greedy search algorithm is presented that achieves almost the same performance as an exhaustive search technique (98% of the time it results in the same schedule) with very low complexity and is applicable for real-time application.

Cross-layer design breaks away from traditional network design where each layer of the protocol stack operates independently. We explore the potential synergies of exchanging information between different layers to support real-time video streaming. In this new approach information is exchanged between different layers of the protocol stack, and end-to-end performance is optimized by adapting to this information at each protocol layer. We discuss key parameters used in the cross-layer information exchange along with the associated cross-layer adaptation. Substantial performance gains through this cross-layer design are demonstrated for video streaming. 1.

We study peer-to-peer multicast streaming, where a source distributes real-time video to a large population of hosts by making use of their forwarding capacity rather than relying on dedicated media servers. Hosts which may disconnect at any time, therefore a robust control protocol is needed to maintain connectivity among peers. This work presents a new peer-to-peer multicast protocol and analyzes the gains that video coding and prioritized packet scheduling at the application layer can bring to the overall streaming performance. A rate-distortion model which predicts endto-end video quality in throughput limited environments is presented and used to determine the over-provisioning necessary to avoid self-inflicted congestion. The video stream transmitted by the source contains H.264 SP and SI frames, which are used to adaptively stop error propagation due to packet loss. Distortion-optimized retransmission requests are issued by receiving hosts to recover the most important missing packets while limiting the induced congestion. Experiments for several hundred hosts simulated in NS-2 illustrate the benefits of our system. We achieve typical end-to-end delays of 1 sec, and a stable video quality with less than 2.5 % of frames lost to playout interruptions. Categories and Subject Descriptors

This paper addresses the problem of streaming packetized media over a lossy packet network to a wireless client, in a rate-distortion optimized way. We introduce an incremental redundancy error correction scheme that combats the effects of both packet loss and bit errors in an end-to-end fashion, without support from the underlying network or from an intermediate base station. The scheme is employed within an optimization framework that enables the sender to compute which packets it should send, out of all the packets it could send at a given transmission opportunity, in order to meet an average transmission rate constraint while minimizing the average end-to-end distortion. Experimental results show that our system is robust and maintains quality of service over a wide range of channel conditions. Up to 8 dB performance gains are registered over systems that are not rate-distortion optimized, at bit error rates as large as 10 .

This paper addresses the problem of streaming packetized media over a lossy packet network, in a rate-distortion optimized way, where the rate-distortion optimization is performed at the receiver. We show how the receiver should compute which packets, if any, to request from the sender, in order to meet a constraint on the transmission rate from the sender to the receiver while minimizing the average end-to-end distortion. Experimental results show that the receiver-driven system has performance within 1 dB of a sender-driven system, while reducing the computational burden at the sender.

Abstract—We consider streaming of video sequences over both constant and variable bit-rate (VBR) channels. Our goal is to enable decoding of each video unit before exceeding its displaying deadline and, hence, to guarantee successful sequence presentation even if the media rate does not match the channel rate. In this work, we will show that the separation between a delay jitter buffer and a decoder buffer is in general suboptimal for VBR video transmitted over VBR channels. We will specify the minimum initial delay and the minimum required buffer for a given video stream and a deterministic VBR channel. In addition, we provide some probabilistic statements in case that we observe a random behavior of the channel bit rate. A specific example tailored to wireless video streaming is discussed in greater details and bounds are derived which allow guaranteeing a certain quality-of-service even for random VBR channels in a wireless environment. Simulation results validate the findings. Index Terms—Receiver buffer, streaming video, variable bit-rate (VBR), wireless video. I.