Transmitting real-time audio or video applications over the Internet is a challenge in the current technology. The motivation for deploying this technology is the reduction in voice communication overheads and the enhancement of services. Voice over Internet Protocol (VoIP) provides improved features like flexible call routing, unified messaging and call center and network multimedia applications which in turn provide reduced costs and improvised services for distance learning, customer support, and remote sales presentations. The integration of voice, video, and data encounters a variable amount of jitter and delay. Typical packet loss ranges from 0 to20 percent and one-way delay from 5 to 500 milliseconds. Reducing jitter delay involves buffering of audio packets at the receiver so that the slower 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 thesis, the existing jitter buffer model was further enhanced by proposing a model to change the codecs from a higher bit rate to a lower bit rate during an established call session thus reducing the packet loss and improving the call performance. A simulation model is shown to support this proposal, leading to the development of a new protocol. Various tests were conducted to analyze the performance of a number of calls and bandwidths by varying one and keeping the other constant.