網際網路與全球資訊網的相關技術的出現為我們的生活帶來相當戲劇性且重要的改變。在許多應用程式中，網路電話備受矚目，並且成為一種新興的社交活動。無論是傳統電信業者或是即將跨入電信市場的業者，提供網路電話的服務是相當緊急而且必要的。
一般來說，網路電話有三個主要的元件所組成：網際網路與傳統電信網路之間的閘道器，高速的網際網路與信號控制系統。閘道器直接接受從傳統電信網路所撥入的電話，同時觸發遠端的閘道器，並且在網際網路上傳送語音資料。當此撥號連接時，此閘道器必須同時為使用者提供全雙工、即時的語音溝通。

在我們的實驗中，我們可以發現到下列幾項特性。舉例來說，當網際網路上的封包發生壅塞的情況時，這個封包會被排在發生壅塞的路遊憩上直到情況改善，我們將這種清況稱做”burst”。因此我們發展了一種緩衝器控制的演算法來避免這種情況的發生，並且盡可能的減低因為封包延遲所造成的遺失率。

在這篇論文中，我們提出一個方法來條者緩衝器的大小來避免封包連續遺失的情況。而調整的最佳時機是在靜音或是子音的時候。因為當我們在這個時候調整緩衝器大小時，人類的聽覺並不容易感覺到聲音被破壞的現象。

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.

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