Abstract
The vast progresses in Internet and WWW related technologies have brought dramatic and fundamental changes to the world that we live. Among the many applications, VoIP has attracted a lot of attention and is regarded as one of the Internet applications that affect how human community. Sooner or later, the VoIP service will be imperative to a greater extent in the telephony market.Generally, three main components of VoIP are: the Internet/PSTN gateway server, the high-speed Internet and the signal and control system. The gateway accepts direct phone dialup from local Internet phone users; it interacts with remote gateway servers and transmits the voice over the Internet (VoIP). In this study we focus on designing an end-to-end adaptation mechanism to deliver these voice packets over the IP networks with unpredictable traffic.We commenced studying the network traffic by conducting a large amount of experiments on TANET. When packet congestion situation occurs, it would be queued in some routers until the traffic jam is removed. Our jitter buffer control algorithm has been developed to prevent this problem so as to decrease the phenomenal of packet loss, especially for burst loss.A scheme of tuning the jitter buffer to adapt the network traffic as well as the ON/OFF activity of voice was proposed. And a real Internet telephony service system with four gateway servers was implemented island wide. The results show that our end-to-end adaptation scheme is workable with acceptable voice quality in TANET environment.