Jitter Buffer Analysis

VoIP is rapidly growing and widely used real-time voice service. On the path through the packet-switched networks, the regularity of VoIP stream is impaired by routing, queuing, scheduling and serialization effects, consequently resulting in loss and delay jitter of packets. Achieving high quality real-time voice requires smoothing the delay jitter at the receiver which is generally done by means of jitter buffer mechanism. Although being an important VoIP element, the exact analysis of the jitter buffer is rather rare. In this paper we model the jitter buffer mechanism and carry out the analysis of its performance. The relations between such quantities as initial playout delay, delay jitter and loss are established. Then, these relations are used together with the voice quality evaluation methodologies MOS and E-model, to optimally choose the controlling parameters of the jitter buffer, in order to increase the perceived quality of the voice. The analytic results developed in this work are applicable for incorporation in playout algorithms to achieve better voice quality.