行動網際網路電話之研製與實現

The Study and Implementation of Mobile Internet Phone

李昆達(Kun-Ta Lee)

指導教授 ： 吳和庭

國立臺北科技大學/機電學院/資訊工程系所/碩士(2005年)

https://doi.org/10.6841/NTUT.2005.00213

摘要

由於通訊技術的進步，網路頻寬的大幅提升，使得網路上可以傳送的資料形式有了重大的變化。在以前，網路上傳送的資料形式，因為受限於頻寛的因素，大部份均以文字檔為主，但由於網路頻寬大幅增加的因素，連帶的使得送的資料形式不在局限傳統的文字檔，像是圖片、語音及影像，也都變的輕而易舉。所以，在網際網路蓬勃發展的趨勢下，傳統的電信網路與數據網路已逐漸整合成以IP為基礎的資訊網路世界，網際網路電話(Internet Phone)的研究、發展與實現，也成為熱門的當紅炸子雞。在網際網路上傳送即時的語音資料只是第一步，未來，更將可以傳送其他即時的多媒體資料，如圖片檔、影片檔，使得人與人的溝通更多彩多姿。
因此，在本論文中，我們利用SIP、RTP及RTCP三種通訊協定為基礎，研製出以IP為基礎的行動網際網路電話系統。首先將利用SIP通訊協定，建立起傳送端與發送端之間的連線，並決定語音資料的初始編碼方式。接著，利用RTP通訊協定，來傳輸即時數位語音資訊資料並且透過RTCP訊號的交換隨時監控即時的網路狀態。因此，在連線中經由SIP訊息的再次交換資料，語音資訊的傳輸量與編碼方式將可隨著行動網路通道傳輸品質的改變而動態調整，那麼傳達的即時語音訊息較不會因網路問題，而造成嚴重失真或延遲的現象。我們以一些開放原始碼做為基礎，再經適當的修改與功能之改善，完成本論文行動網際網路電話系統之實現。

關鍵字

SIP ； RTP ； RTCP ； Mobile ； 行動網際網路 ； 網路電話

並列摘要

With the advance of the communication transmission technology, the network bandwidth has expanded significantly for the past few years. As a result, the data carried by the network transmission media is no longer limited to text based data format anymore. Instead, we have witnessed the explosive growth of multimedia applications with rich data formats, such as images, audio and video data. In addition, thanks to the widespread deployment of the internet, the traditional telecom network and the datacom network has converged into the IP-based infocom world gradually. Therefore, the study and development of Voice over IP (VoIP) technology has attracted wide attention recently. It is expected that with the mature development of Internet phone technology, the Internet will enable more diverse real time multimedia applications.
Therefore, in this thesis, we study and implement the IP based mobile Internet Phone, based upon the following three communication protocols: SIP, RTP, and the RTCP. The SIP protocol, a light weight signaling protocol, is used for the connection establishment between the caller and the callee. The initial codec format is determined via the SIP signaling exchange. The real time packet switched digital voice data are then carried as the RTP protocol data units over UDP datagrams. In conjunction with the RTP packets, the periodical RTCP packets contain both the sender and receiver reports. These RTCP packets collect real time statistics that can provide an excellent indicator of current network status. It is known that the channel quality of wireless environments changes very quickly and dynamically. Therefore, through the re-activation of SIP protocol during the connection, the mobile IP phone system can thus adjust the transmission rate and codec format of the voice data accordingly, if necessary. The QoS of mobile voice data can thus be sustained. We have modified a few related open source libraries and then expand their functions for realizing such a mobile internet phone system.

並列關鍵字

SIP ； RTP ； RTCP ； Mobile ； Internet ； Internet Phone

參考文獻

[3] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, “SIP: Session Initiation Protocol,” Internet Engineering Task Force, RFC 2543, March 1999.

[5] S. Donovan, “The SIP INFO Method,” Internet Engineering Task Force, RFC 2976, October 2000.

[7] J. Rosenberg and H. Schulzrinne, “Reliability of Provisional Responses in Session Initiation Protocol (SIP),” Internet Engineering Task Force, RFC 3262, June 2002.

[8] J. Rosenberg and H. Schulzrinne, “Session Initiation Protocol (SIP): Locating SIP Servers,” Internet Engineering Task Force, RFC 3263, June 2002.

[1] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications,” Audio-Video Transport Working Group, RFC 1889, January 1996.