Abstract
This paper presents new speech coding issues and algorithms involved in IP telephony. To develop an efficient system of adaptive voice over IP (VoIP), in fact, besides traditional networking aspects, a series of speech processing issues need to be carefully considered. More specifically, two important aspects of discontinuous transmission are dealt with: the impact of the VAD on source throughput and the need for an efficient system of comfort noise. In addition we propose a variable rate, toll quality CS-ACELP coder that uses coding modes compatible with the three 6.4, 8, and 11.8 kbit/s coding schemes standardised by ITU-T in G.729. In particular, the algorithm presents 4 coding categories, with an average bit rate ranging between about 3 and 8 kbit/s, that adapt the rate to changes in network conditions.
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Beritelli, F., Casale, S., Francese, M., Ruggeri, G. (2001). Intrastandard Hybrid Speech Coding for Adaptive IP Telephony. In: Marsan, M.A., Bianco, A. (eds) Quality of Service in Multiservice IP Networks. QoS-IP 2001. Lecture Notes in Computer Science, vol 1989. Springer, Berlin, Heidelberg. https://doi.org/10.1007/3-540-44554-4_12
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DOI: https://doi.org/10.1007/3-540-44554-4_12
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