A New Buffer Algorithm for Speech Quality Improvement in VoIP Systems

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).

[1]  Mark Carson,et al.  NIST Net: a Linux-based network emulation tool , 2003, CCRV.

[2]  JongWon Kim,et al.  Adaptive delay concealment for Internet voice applications with packet-based time-scale modification , 2001, SPIE Optics East.

[3]  Masayuki Murata,et al.  Adaptive Playout Buffer Algorithm for Enhancing Perceived Quality of Streaming Applications , 2004, Telecommun. Syst..

[4]  Lingfen Sun,et al.  New models for perceived voice quality prediction and their applications in playout buffer optimization for VoIP networks , 2004, 2004 IEEE International Conference on Communications (IEEE Cat. No.04CH37577).

[5]  Donald F. Towsley,et al.  Packet audio playout delay adjustment: performance bounds and algorithms , 1998, Multimedia Systems.

[6]  Henning Schulzrinne,et al.  Adaptive playout mechanisms for packetized audio applications in wide-area networks , 1994, Proceedings of INFOCOM '94 Conference on Computer Communications.

[7]  Costas S. Xydeas,et al.  Model-based packet loss concealment for AMR coders , 2003, 2003 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2003. Proceedings. (ICASSP '03)..

[8]  JongWon Kim,et al.  Adaptive delay concealment for Internet voice applications with packet based time-scale modification , 2001, 2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221).

[9]  Bernd Girod,et al.  Adaptive playout scheduling and loss concealment for voice communication over IP networks , 2003, IEEE Trans. Multim..