This paper presents a dynamic adaptation algorithm of joint source-channel code rate for enhancing voice transmission over LTE network. In order to assess the speech quality, we use the Wideband (WB) E-model. In this model, both end-to-end delay and packet loss are taken into account. The goal of this paper is to find out the best suboptimal solution for improving voice traffic over LTE network with some constraints on allowed maximum end-to-end delay, allowed maximum packet loss and minimum required bandwidth. The best suboptimal choice is channel code rate corresponding to each mode of the AMR-WB codec that minimizes redundant bits generated by channel coding with an acceptable MOS reduction. Besides, this algorithm can be integrated with rate control in AMR-WB codec to offer the required mode of LTE network. Our results show that the MOS degradation is not significant, but the percent of reduced redundant bits to be very considerable. This will requires less bandwidth, thus, more mobile users can be served. The algorithm has simple computational operations, it can be applied to real-time voice communications.