Wireless Personal Communications

, Volume 45, Issue 2, pp 189-207

First online:

A New Buffer Algorithm for Speech Quality Improvement in VoIP Systems

  • Zizhi QiaoAffiliated withMotorola Inc.Signal Processing & Multimedia Communications Group, University of Plymouth Email author 
  • , Ramesh K. VenkatasubramanianAffiliated withMotorola Inc.
  • , Lingfen SunAffiliated withSignal Processing & Multimedia Communications Group, University of Plymouth
  • , Emmanuel C. IfeachorAffiliated withSignal Processing & Multimedia Communications Group, University of Plymouth

Rent the article at a discount

Rent now

* Final gross prices may vary according to local VAT.

Get Access


Jitter buffer plays an important role in Voice over IP (VoIP) applications because it provides a key mechanism for achieving good speech quality to meet technical and commercial requirements. The main objective of this paper is to propose a new, simple-to-use jitter buffer algorithm as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance, in terms of enhanced user-perceived speech quality and reduced end-to-end delay. Supported by signal processing features, the new algorithm, the so-called Play Late Algorithm, alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. The results show that the new algorithm achieves the best performance under different network conditions when compared to conventional static and adaptive jitter buffer algorithms. The results reported here are based on live tests and emulated network conditions on real mobile phone prototypes. The mobile phone prototypes use AMR codec and support full IP/UDP/RTP stack with IPSec function in some of the tests. The method for perceived speech quality measurement is based on the ITU-T standard for speech quality evaluation (PESQ).


Jitter buffer Voice over IP Quality of service (QoS) Perceived speech quality PESQ AMR codec