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This paper proposes an algorithm for the adaptive adjustment of the transmission rate of VoIP sources based on the voice quality estimated at the receptor. This adjustment is achieved through the appropriate use of differing voice codecs as the conditions of the network change, with the aim at maintaining an efficient utilization of the available resources. In order to validate our proposal, we have made an effort to simulate more realistic VoIP calls using sources that follow Brady's model of human conversations. We investigate the effect of the proposed model on the aggregate network traffic and compare the results with existing related work, most of which employ simplified models. Simulation results show that the proposed algorithm makes efficient use of the available bandwidth, achieving superior performance in comparison to similar works from the literature.Este artigo propõe um algoritmo para o ajuste adaptativo da taxa de transmissão de fontes VoIP com base na qualidade da voz estimada pelo receptor. O ajuste é realizado através da utilização de diferentes codificadores, dependendo das condições da rede, visando fazer uso de forma eficaz dos recursos disponíveis. Na validação de nossa proposta, realizamos simulações buscando modelar os ambientes de chamadas VoIP de uma forma realística. Para isso utilizamos fontes VoIP baseadas no modelo de Brady de uma conversação humana. Os efeitos da modelagem e do algoritmo proposto sobre o tr´fego agregado de uma rede de dados foram analisados, através da comparação dos resultados obtidos com trabalhos relacionados que utilizam modelos simplificados. Esses resultados mostram que o algoritmo proposto consegue aproveitar de forma eficaz a banda disponível, e apresenta um desempenho superior a trabalhos similares da literatura.