Adaptable technique for recovering lost internet audio packets

With rapid progress in both computers and networks, real-time multimedia applications are now possible on the Internet. Since the Internet was designed to support traditional applications, multimedia applications on the Internet often suffer from unacceptable delay, jitter and data loss. Among these, data loss has the largest impact on quality. When the retransmission technique is used to recover lost audio packets, more obstacles, like the delay time resulted from lost packets retransmission, network traffics overloading, and buffer management, are faced. All above demonstrated problems are due to the real-time communication features. In this paper, we demonstrate a new technique that studies the audio packet's priority and it's effect on the transmission process. Hence, our technique arranges the audio packets depending on there importance. So, before deciding to retransmit a packet, we should determine if this packet could be recovered by other suitable technique or not. Hence, the number of retransmitted packets is decreased. Therefore, all above problems may be calmed. Some additional parameters, like a buffer management and a packet position are taken in the consideration. These parameters help us in selecting an ideal recovery technique for lost audio packets. Finally, NS2 is used to build a simulated environment for testing our technique.