Adaptive Codec Mode Assignment for Voice over Ip with Amr Speech Codec

In this paper, we proposed a new VoIP system that employs an AMR speech codec which can vary its bit-rate from 12.2 kbits/s to 4.75 kbits/s on a frame basis depending on the network channel conditions. For monitoring of channel conditions every frame, we calculated interarrival jitter using the timestamp information in the RTP header. We then developed the AMR speech codec mode assignment algorithms for VoIP using the estimated channel conditions as follows: 1) simply divide the interarrival jitter range linearly into 8 state, 2) change the range of each codec mode every RTCP packet interval adaptively depending on the long-term estimated channel conditions using both ‘cumulative number of packets lost’ parameter in the header of RTCP packet and distribution characteristics of the assigned codec modes during that period. Experimental results demonstrate the superiority of the VoIP with an AMR speech codec to the conventional VoIP systems.