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 buffered, and their playout delayed at the destination host in order to compensate for the variable network delays. The authors 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. They evaluate the playout algorithms using experimentally-obtained delay measurements of audio traffic between several different Internet sites. Their results indicate that an adaptive algorithm which explicitly adjusts to the sharp, spike-like increases in packet delay which were observed in the traces can achieve a lower rate of lost packets for both a given average playout delay and a given maximum buffer size. >

Adaptive playout mechanisms for packetized audio applications in wide-area networks

In packet audio applications, packets are buffered at a receiving site and their playout delayed in order to com- pensate for variable network delays. In this paper, we con- sider 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 al- ready 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 in- formation, together with a "delay spike" detection algorithm based on (but extending) our earlier work, is used to dy- namically 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.

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Packet audio playout delay adjustment: performance bounds and algorithms

A network data stream synchronization method and system are disclosed. Operating delay and loss parameters are accepted and a data stream buffer is initialized. The network delay experienced by a data packet is determined and a buffer delay is imposed to provide a fixed end-to-end delay. Packets arriving too late to be played within the fixed end-to-end delay are discarded. Network delay models may be updated and clock drifts may be detected and compensated for using historical buffer data.

Synchronization system for networked multimedia streams

Presents an adaptive flow synchronization protocol that permits synchronized delivery of data to and from geographically distributed sites. Applications include inter-stream synchronization, synchronized delivery of information in a multisite conference, and synchronization for concurrency control in distributed computations. The contributions of this protocol in the area of flow synchronization are the ability to synchronize over arbitrary topologies, the introduction of an adaptive synchronization delay, the flexibility to maintain multiple synchronization groups, and the use of a modular architecture that permits the application to tailor synchronization calculations to its service requirements. The authors take advantage of network protocols capable of maintaining network clock synchronization in the millisecond range. >

Flow synchronization protocol

The transmission of real-time streams over best effort networks has been an interesting research area for over a decade. An important objective of the research community has been to devise methods that cope with the variations of the network delay-also called delay jitter-that are an inherent characteristic of best effort networks. Jitter destroys the temporal relationships between periodically transmitted media units, that constitute a real-time media stream, thus hindering the comprehension of the stream. Playout adaptation algorithms undertake the labor of the temporal reconstruction of the stream, which is sometimes referred to as the restoration of its' intrastream synchronization quality. This article surveys the work in the area of playout adaptation, aiming to concisely organize ideas that have been presented in isolation and identify the main points of differentiation among different schemes. The survey, discusses issues related to timing information, handling of late media units, quality evaluation metrics, and adaptation to changing delay conditions.

Intrastream synchronization for continuous media streams: a survey of playout schedulers

An algorithm for voice synchronization for packet switching networks is presented. The algorithm has been tested both in simulation and on a real network. The algorithm runs on the TRAME packet switching network for both the Vocoder and CELP DoD voice coding standards. Some results of these tests are presented. Some details of the algorithm development and implementation are given as well. >

J.M. Selga

Papers

Voice synchronization in packet switching networks