Packet audio playout delay adjustment: performance bounds and algorithms
Multimedia Systems
Voice over IP performance monitoring
ACM SIGCOMM Computer Communication Review
Assessing Voice Quality in Packet-Based Telephony
IEEE Internet Computing
Delay reduction techniques for playout buffering
IEEE Transactions on Multimedia
Frequency-domain adaptive algorithm for network echo cancellation in VoIP
EURASIP Journal on Audio, Speech, and Music Processing - Intelligent Audio, Speech, and Music Processing Applications
WSEAS TRANSACTIONS on COMMUNICATIONS
| Hi-index | 0.24 |
In order to transfer voice or some other application requiring real-time delivery over a packet network, we need a de-jitter buffer to eliminate delay jitters. An important design parameter is the depth of the de-jitter buffer since it influences two important parameters controlling voice quality, namely voice-path delay and packet loss probability. In this paper, we propose and study several schemes for optimally adjusting the depth of the de-jitter buffer. In addition to de-jitter-buffer depth adjustments within a call, the initial value and rates of changes of the de-jitter buffer depth are allowed to depend on the class of the call and are adaptively adjusted (upwards or downwards) for every new call based on voice-path delay and packet loss probability measurements over one or more previous calls. Parameter adjustments are geared towards either (a) minimizing voice-path delay while maintaining a packet loss probability objective, or (b) maximizing R-factor, an objective measure of voice quality that depends both on the voice-path delay and the packet loss probability. Using simulation models and measured packet delay traces, it is shown that adaptive schemes perform better than static ones and adaptive schemes with learning perform better than ones without learning.