以即時協定為基礎的網際網路電話研究

Abstract

現在網際網路是電信的一部份，以後電信將成為網際網路的一部份。電腦電話閘道器(或稱為網際網路電話閘道器)是電信與網際網路的溝通橋樑，通常使用電腦加裝電話語音卡，成為電腦電話閘道器。本論文首先提出兩種利用每部個人電腦通常就具備的數據機、聲霸卡和網路卡，來完成電腦電話閘道器的功能架構。 雖然寬頻網路時代的來臨，有助於在網際網路傳送的語音品質的提昇，但網路分享的網路特性，更多的多媒體與語音資料，在同時使用有限的頻寬，因此仍需要有效地使用頻寬。目前利用雜訊降低、回音消除、語音壓縮、動態撥放暫存(Adaptive playout buffer)及即時協定(RTP)等技術來提昇語音在網際網路傳送的品質，其中語音壓縮技術是能否有效運用頻寬的關鍵，而且壓縮技術之發展已臻成熟。 經由網際網路電話軟體SpeakFreely之實測結果可知，語音封包的遺失發生於頻寬不足，極長的語音延遲(spike)發生於網路擁塞。本論文提出由即時協定的語音封包接收情形，計算出目前之網路頻寬的數學公式，以便選用最適合的語音壓縮技術；並提出改善的spike偵側公式，於spike發生的最初期，立即改用較高壓縮率的語音壓縮方法。藉spike和頻寬不足之偵測及解決的演算法，來改善網際網路電話的語音品質。

Now the internet is a part of the telecommunication, but in the future telecommunication will be a part of the internet. The computer telephony gateway (or called internet telephony gateway) is the bridge between telecommunication and internet. Usually using a computer added a telephony interface card becomes a computer telephony gateway. The thesis first offers two architecture of the internet telephony gateway using modems, a sound card and a network interface card that every personal computer already has. Although the age of broadband network comes, there are more multimedia and voice data share the finite bandwidth at the same time, it still need to use bandwidth efficiently with the characteristic of network sharing. The technology of noise reduction, echo canceling, voice compression, adaptive playout buffer and Real Time Protocol (RTP) are used to improve the voice quality in internet telephony. And the technology of voice compression is the key to use bandwidth efficiently and developed maturely. With the internet telephony software “SpeakFreely” we experimented and found out that the packet loss took place in insufficient bandwidth and the spike took place in network jam. The thesis offers a mathematic formula to measure the network bandwidth with RTP voice packets, then chose the optimal voice compression, and offers an improved spike detection formula to early detect the spike and change to the more compact compression method at the beginning of the spike. With the algorithm for detection and solution of spike/bandwidth insufficient, the voice quality of internet telephony is improved.