Influence of codecs on adaptive jitter buffer algorithm

Abstract

Transmitting real-time audio or video applications over the Internet is a challenge in the current networking technology. The motivation for deploying real-time applications includes the reduction in voice communication overheads and the enhancement of services. The integration of voice, video, and data encounters a variable amount of jitter and delay. Typically packet loss ranges from 0% to 20% and one-way delay from 5 to 500 milliseconds [1] [2]. Reducing jitter delay involves buffering of audio packets at the receiver so that the packets arrive sequentially on time at the destination. Adaptive jitter buffering at the receiver improves the quality of voice connections on the Internet. In this study, a simulation model was proposed to further enhance the existing jitter buffer model to change the voice codecs dynamically. Voice codecs were changed from higher bit rate to lower bit rate during an established call session based on jitter buffer value. The proposed model reduced the packet loss thereby improving the call performance during the on-going call session.