| 摘要: | 在網際網路快速發展的年代，在VoIP( Voice over Internet Protocol )低成本通話的優勢及對通話品質的大幅改善下，VoIP 已被廣泛應用於商業或個人環境。但是這些VoIP 應用軟體如SKYPE 必須事先安裝客戶端軟體在個人電腦上才能使用，而且繁雜安裝設定步驟對不熟悉電腦操作的使用者來說，增加使用上的困擾同時在公眾環境的電腦上也不方便事先安裝軟體下形成使用不便。更嚴重的是VoIP 軟體使用SIP (Session Initiation Protocol)標準協定，該協定針對建立雙方通話使用即時傳輸協定RTP (Real-Time Protocol) 通訊中對NAT (Network Address Translation) 穿越的問題並沒有完善的處理，容易發生語音單向通話或是雙方無法建立通話的狀況。 本研究針對上述問題，提出結合網頁瀏覽操作的方便性與普遍性、以及IAX2開放通訊協定特性，開發出一套網頁電話介面（稱為WebCall 系統），免除複雜安裝手續的困擾與負擔。同時因IAX2 中，將信令控制訊息和語音資料流使用同一個網路通訊埠傳輸，可改善SIP 中對NAT 穿透所產生的問題。為了避免在錯誤率很高的網路環境下，可能會因遺失太多封包而影響通話品質，本研究運用冗餘封包保護技術來提高傳輸正確率，並選用免授權金與節省頻寬的GSM codec 語音壓縮格式來減輕傳輸語音封包負擔。我們經由實驗發現以語音平均意見得分 (Mean Opinion Score, MOS) 與R-Factor 語音品質數據值的來衡量，在網路環境當封包延遲100 毫秒和封包遺失10% 下，WebCall 系統與一般VoIP系統的比較明顯改善15%，說明以IAX2 為基礎之網頁電話相較於傳統SIP 軟體網絡電話有明顯改善頻寬傳輸效率，有助於雙方通話的語音品質。 In an age of fast-developing Internet with the advantage of low communication costof VoIP and improvement in communication quality, VoIP has been widely applied in business or personal environment. However, the VoIP application software package like SKYPE has to be installed in advance in the PC at the customer site before using;besides, the complicated installation and configuration steps may beset the users who are not familiar to the computer operation; it will cause some trouble in usage. In the same time, for the computers in a public environment, pre-installation of software packages is not that convenient, this also leads to usage inconvenience. What could be more serious is that VoIP adopts SIP standard protocol; the said protocol does not offer sound solution to the problem of NAT penetration regarding the mutual call set-up for the communication which adopts RTP (Real Time Protocol), it thus leads to single-way voice communication situation or mutual communication can not be set up. In this study, the author has focused on the above-mentioned problems and proposed a solution of integrating the convenience and popularity of the web browsing operation as well as the open feature of IAX2 protocol, and, developed a Web Phone Interface (It is named WebCall System). It frees the user from the perplexity and burden of the complicated installation procedures. Besides, in IAX2, both the signaling control message and media stream use the same network communication port to transmit, it could ameliorate the problems caused by NAT penetration in SIP. In order to avoid the possibility of impacting the communication quality due to missing too many packets under the network environment of high error rate, in this study, we have made use of redundant packet protection technology to enhance the transmission accuracy and selected Royalty-free and bandwidth-saving of GSM Codec Voice Compression format to alleviate the load of voice packet transmission. We have found out through the experiment that when using the Mean Opinion Score and the R-Factor Voice Quality Figures for measurement, under the environment of packet delay of 100 ms and packet loss of 10%, the WebCall system demonstrates obvious 15% improvement comparing to the common VoIP system; it shows that an IAX2-based Webpage-Phone presents obvious improvement in bandwidth transmission efficiency in comparison to the traditional SIP Soft-Phone, this great helps the voice quality of mutual communication. |
