Parametric mixing for centralized VoIP conferencing using ITU-T recommendation G.722.2
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Excessive end to end delay or latency in a VoIP network is one of the major drawbacks of packetized voice communication. The two main components of end to end delay are codec processing and propagation delay. Codec processing delay becomes a significant factor in centralized VoIP conferencing where voice packets, originating from the participants, are sent to a central unit or bridge to be combined using tandem mixing.This thesis investigates a novel way based on the G.722.2 codec of mixing the packets at the central unit that reduces algorithmic complexity and therefore delay. The parameters used to represent the speech, LPCs, pitch lags, fixed codebook, and gains, are extracted from the encoded bit stream, mixed, and re-encoded instead of full decoding, mixing, and then re-encoding of the speech signals.This parametric mixing reduces the bridge complexity by up to 85 % while still retaining acceptable speech quality, 3.7 MOS on average, as shown by simulations.
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Copyright © 2006 the author(s). Theses may be used for non-commercial research, educational, or related academic purposes only. Such uses include personal study, research, scholarship, and teaching. Theses may only be shared by linking to Carleton University Institutional Repository and no part may be used without proper attribution to the author. No part may be used for commercial purposes directly or indirectly via a for-profit platform; no adaptation or derivative works are permitted without consent from the copyright owner.
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- 2006
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