Proceedings of the 3rd International Conference on Measurement of Speech and Audio Quality in Networks, Prague, Czech Republic, May 2004.
Abstract: In packetized speech transmission, end–to–end delay can vary, even over short timescales. Estimating the resulting speech delay histories is critical to diagnostic and quality estimation efforts. We present a new bottom–up algorithm for estimating time–varying speech delays. The bottom–up approach is well–suited to real–time implementation. The algorithm works with very low–rate codecs as well as the higher–rate codecs that are more common in VoIP applications. We describe the new algorithm in some detail and provide descriptions of the databases and techniques used to develop and test the new algorithm.
For technical information concerning this report, contact:
Stephen D. Voran
Institute for Telecommunication Sciences
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