Study on the application of an AMR speech codec to VoIP
Jeong Wook Seo, Se Jeong Woo, Keun Sung Bae · 2002
Degradation of speech quality caused by packet loss of voice traffic is still one of critical technical barriers of the VoIP system. We propose a new VoIP system that can adapt transmission bit rate flexibly to network conditions to reduce packet loss. In order to determine the transmission bit rate depending upon the network conditions on a frame basis, we use the time-stamp parameter in the RTP of the H.323 protocol. Experimental results demonstrate that the proposed system is very promising to reduce packet loss that leads to improvement of speech quality.