Anti-run-dry algorithm for optimal control of playoutbuffers
ISICT '03 Proceedings of the 1st international symposium on Information and communication technologies
Switching between Fixed and Call-Adaptive Playout: A Per-Call Playout Algorithm
IEEE Internet Computing
A moving average predictor for playout delay control in VoIP
IWQoS'03 Proceedings of the 11th international conference on Quality of service
QoS-based optimal adaptive playout-buffer scheduling using the packet arrival distribution
MILCOM'09 Proceedings of the 28th IEEE conference on Military communications
Optimal adaptive voice smoother with Lagrangian multiplier method for VoIP service
MUSP'06 Proceedings of the 6th WSEAS international conference on Multimedia systems & signal processing
Adaptive voice smoothing with optimal playback delay based on the ITU-T e-model
EUC'05 Proceedings of the 2005 international conference on Embedded and Ubiquitous Computing
Adaptive voice smoother with optimal playback delay for new generation VoIP services
EUC'05 Proceedings of the 2005 international conference on Embedded and Ubiquitous Computing
Adaptive VoIP smoothing of pareto traffic based on optimal e-model quality
PCM'05 Proceedings of the 6th Pacific-Rim conference on Advances in Multimedia Information Processing - Volume Part II
| Hi-index | 0.00 |
In a typical real-time voice application, voice packets are produced at deterministically-spaced time intervals. In the network they encounter a variable amount of delay that changes the deterministic time intervals. A receiving host can employ a buffer to delay the playout of the voice packets in order to reconstruct the original timing. Adaptive techniques can perform continuous estimations of the network delays and dynamically adjust the buffering delay at the beginning of each talkspurt. Such adjustments are usually undetectable by the human listener. This research develops a new, adaptive "gap-based" algorithm that can be tuned for both end-to-end delay and packet loss to satisfy a user-desired tolerance. This new gap based algorithm adapts the buffering delay based on historical information of arrival and playout times of received voice packets in the previous talkspurt. A simulation study shows that the new gap based algorithm can reduce delay by 10% when compared with existing methods.