Delay reduction techniques for playout buffering

Abstract

Receiver synchronization of continuous media streams is required to deal with delay differences and variations resulting from delivery over packet networks such as the Internet. This function is commonly provided using per-stream playout buffers which introduce additional delay in order to produce a playout schedule which meets the synchronization requirements. Packets which arrive after their scheduled playout time are considered late and are discarded. In this paper, we present the Concord algorithm, which provides a delay-sensitive solution for playout buffering. It records historical information and uses it to make short-term predictions about network delay with the aim of not reacting too quickly to short-lived delay variations. This allows an application-controlled tradeoff of packet lateness against buffering delay, suitable for applications which demand low delay but can tolerate or conceal a small amount of late packets. We present a selection of results from an extensive evaluation of Concord using Internet traffic traces. We explore the use of aging techniques to improve the effectiveness of the historical information and hence, the delay predictions. The results show that Concord can produce significant reductions in buffering delay and delay variations at the expense of packet lateness values of less than 1%.Index Terms-Multimedia stream synchronization, playout buffering.

A New Packet Scheduling Algorithm for Real-Time Multimedia Streaming

“…We gave description of jitter in previous sections. In the multimedia applications, calculation of playout buffer is done according to maximum jitter [20]. A playout buffer is a holding area for packets which will be played in the future.…”

An empirical evaluation of VoIP playout buffer dimensioning in Skype, Google talk, and MSN Messenger

“…The reported simulation results show that this approach improves the tradeoff between buffer delay and packet loss significantly. The approaches proposed in [9,12] further extended [10,11] by adjusting the buffer size within a speech burst. The objective is to ensure that the playout buffer adapts to varying network conditions more quickly, and thereby improve the conversation quality of VoIP calls.…”

“…Several VoIP playout buffer dimensioning algorithms have been proposed [9][10][11][12]. Most of them adjust the buffer size based on a linear combination of network delay and jitter.…”

An adaptive playout algorithm with delay spike detection for real-time VoIP

“…Three variations are presented in [45]. In Algorithm 1, each bin count is multiplied by a constant aging factor and the bin containing the delay of the current packet is then incremented by one.…”