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Speech quality prediction in VoIP concatenating multiple Markov-based channels

Lookup NU author(s): Iban Lopetegui, Emeritus Professor Rolando Carrasco, Professor Said Boussakta


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The cross interaction of technologies has become standard practice to add more constraints to Voice over Internet Protocol sessions (VoIP). In this paper, a self developed end to end VoIP simulator to predict speech quality is presented. This system has been validated with a real test bench and can provide up to three audio compressors and two channel status at a time. Simulator response is tested through a Mean Opinion Score (MOS) and compared to ITU-Ts G.107 E-model speech quality predictor. High packet loss rate limitation on the E-model is solved by our proposed new parameter. In addition, a methodology to extend this predictor for multiple concatenated channels has been tested and proved to be successful; results in double channels tests range between 1.86R and 8.35R error rate. The methodology is a useful speech quality predictor for design and management purposes; VoIP gateways can take advantage of channels and codec information to guarantee an specific quality of service to end users. © 2010 IEEE.

Publication metadata

Author(s): Lopetegui I, Carrasco R, Boussakta S

Publication type: Conference Proceedings (inc. Abstract)

Publication status: Published

Conference Name: 6th Advanced International Conference on Telecommunications (AICT)

Year of Conference: 2010

Pages: 226-230

Publisher: IEEE


DOI: 10.1109/AICT.2010.46

Library holdings: Search Newcastle University Library for this item

ISBN: 9780769540214