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Recent interest in supporting packet-audio applications over wide area networks has been fueled by the availability of low-cost, toll-quality workstation audio and the demonstration that limited amounts of interactive audio can be supported by today's Internet. In such applications, received audio packets are bu ered, and their playout delayed at the destination host in order to compensate for the variable network delays. In this paper we investigate the performance of four different algorithms for adaptively adjusting the playout delay of audio packets in an interactive packet-audio terminal application, in the face of such varying network delays. We evaluate the playout algorithms using experimentally-obtained delay measurements of audio tra c between several different Internet sites. Our results indicate that an adaptive algorithm which explicitly adjusts to the sharp, spike-like increases in packet delay which we observed in our traces can achieve a lower rate of lost packets for both a given average playout delay and a given maximum buffer size.

This paper describes current problems found with audio applications over the MBONE (Multicast Backbone) , and investigates possible solutions to the most common one - packet loss. The principles of packet speech systems are discussed, and how the structure allows the use of redundancy to design viable solutions to the problem. The paper proposes the use of synthetic speech coding algorithms (vocoders) to provide redundancy, since the algorithms produce a very low bit-rate stream, which only adds a small overhead to a packet. Preliminary experiments show that normal speech repaired with synthetic quality speech is intelligible, even at very high loss rates. Introduction The application of this work is multimedia conferencing over the MBONE (Multicast Backbone), an experimental overlay network of the Internet. The work has arisen from experiences in multi-way multimedia conferencing in Project MICE (Multimedia Integrated Conferencing for Europe) [1], is currently applied in Project ReL...

We survey a number of packet-loss recovery techniques for streaming audio applications operating using IP multicast. We begin with a discussion of the loss and delay characteristics of an IP multicast channel and from this show the need for packet loss recovery. Recovery techniques may be divided into two classes: senderand receiver-based. We compare and contrast several sender-based recovery schemes: forward error correction (both media specific and media independent) interleaving and retransmission. In addition a number of error concealment schemes are discussed. We conclude with a series of recommendations for repair schemes to be used, based on application requirements and network conditions. 1 Introduction The development of IP multicast and the Internet multicast backbone has led to be emergence of a new class of scalable audio/video conferencing applications. These are based on the lightweight sessions model [11, 17] and provide efficient multi-way communication which scales fr...

In packet audio applications, packets are buffered at a receiving site and their playout delayed in order to compensate for variable network delays. In this paper, we consider the problem of adaptively adjusting the playout delay in order to keep this delay as small as possible, while at the same time avoiding excessive “loss ” due to the arrival of packets at the receiver after their playout time has already passed. The contributions of this paper are twofold. First, given a trace of packet audio receptions at a receiver, we present efficient algorithms for computing a bound on the achievable performance of any playout delay adjustment algorithm. More precisely, we compute upper and lower bounds (which are shown to be tight for the range of loss and delay values of interest) on the optimum (minimum) average playout delay for a given number of packet losses (due to late arrivals) at the receiver for that trace. Second, we present a new adaptive delay adjustment algorithm that tracks the network delay of recently received packets and efficiently maintains delay percentile information. This information, together with a “delay spike ” detection algorithm based on (but extending) our earlier work [RKTS94], is used to dynamically adjust talkspurt playout delay. We show that this algorithm outperforms existing delay adjustment algorithms over a number of measured audio delay traces and performs close to the theoretical optimum over a range of parameter values of interest.

In this paper, we present an ecient Multiple Description Coding (MDC) technique to achieve robust communication over unreliable channels such as a lossy packet network. We first model such unreliable channels as erasure channels and then we present a MDC system using polyphase transform and selective quantization to recover channel erasures. Different from previous MDC work, our system explicitly separates description generation and redundancy addition which greatly reduces the implementation complexity specially for systems with more than two descriptions. Our system also realizes a Balanced Multiple Description Coding (BMDC) framework which can generate descriptions of statistically equal rate and importance. This property is well matched to communication systems with no priority mechanisms for data delivery, such as today's Internet.

Abstract—Network performance evaluation through traditional packetlevel simulation is becoming increasingly difficult as today’s networks grow in scale along many dimensions. As a consequence, fluid simulation has been proposed to cope with the size and complexity of such systems. This study focuses on analyzing and comparing the relative efficiencies of fluid simulation and packet-level simulation for several network scenarios. We use the “simulation event ” rate to measure the computational effort of the simulators and show that this measure is both adequate and accurate. For some scenarios, we derive analytical results for the simulation event rate and identify the major factors that contribute to the simulation event rate. Among these factors, the “ripple effect ” is very important since it can significantly increase the fluid simulation event rate. For a tandem queueing system, we identify the boundary condition to establish regions where one simulation paradigm is more efficient than the other. Flow aggregation is considered as a technique to reduce the impact of the “ripple effect ” in fluid simulation. We also show that WFQ scheduling discipline can limit the “ripple effect”, making fluid simulation particularly well suited for WFQ models. Our results show that tradeoffs between parameters of a network model determines the most efficient simulation approach. Keywords—fluid simulation, performance evaluation, traffic model I.

This memorandum is a companion document to the current version of the RTP protocol specification draft-ietf-avt-rtp-*.ftxt,psg. It discusses protocol aspects of transporting real-time services (for example, voice or video) over packet-switched networks such as the Internet. It compares and evaluates design alternatives for a real-time transport protocol, providing rationales for the design decisions made for RTP. Also covered are issues of port assignment and multicast address allocation. An appendix provides a comprehensive glossary of terms related to multimedia conferencing. This document is a product of the Audio-Video Transport working group within the Internet Engineering Task Force. Comments are solicited and should be addressed to the working group's mailing list at rem-conf@es.net and/or the author(s). INTERNET-DRAFT draft-ietf-avt-issues-02.ps May 9, 1994 Contents 1 Introduction 4 2 Goals 7 3 Services 9 3.1 Control and Data : : : : : : : : : : : : : : : : : : : : : : : : ...

Weinvestigate the interaction of congestion control with the partitioning of source information into components of varying importance for variable bit rate packet voice and packet video. High priority transport for the more important signal components results in substantially increased objective service quality. Using a Markovchain voice source model with simple PCM speech encoding and a priority queue, simulation results show a signal-to-noise ratio improvementof45dBwithtwo priorities over an unprioritized system. Performance is sensitive to the fraction of traffic placed in eachpriority, and the optimal partition depends on network loss conditions. When this partition is optimized dynamically, quality degrades gracefully over a wide range of load values. Results with DCT encoded speech and video samples show similar behavior. Variations are investigated such as further partition of low priority information into multiple priorities. A simulation with delay added to represe...

Real-time audio and video data streamed over unreliable IP networks, such as the Internet, may encounter losses due to dropped packets or late arrivals. This paper reviews error-concealment schemes developed for streaming realtime audio and video data over the Internet. Based on their interactions with (video or audio) source coders, we classify existing techniques into source coder-independent schemes that treat underlying source coders as black boxes, and source coder-dependent schemes that exploit coder-specific characteristics to perform reconstruction. Last, we identify possible future research directions. 1. Introduction Increases in bandwidth and computational speed lead to growing interests in real-time audio and video transmissions over the Internet. In the Internet, packets carrying real-time data may be dropped or arrive too late to be useful because the Internet is a packet-switched, best-effort delivery service, with no guarantee on the quality of service (QoS). Traditi...