在網路電話語音封包傳送的網際網路中，語音通話過程中如何維持穩定的通話服務品質(Quality of Service, QoS)與有效分散架構負載流量是一項重要的議題。有鑑於 SIP RFC3602 的標準中，並未指明使用何種協定 (Protocol) 來讓用戶端 (User Agent, UA)獨佔不同網路特性的資源頻寬，本論文針對此不足，在考量SIP協定網路特性及網路電話架構應用下，提出一個整合資源預留協定之SIP機制，讓通話者通話QoS達成穩定。這個整合機制由SIP整合了整合式網路的RSVP (Reservation service Protocol) 與差異式網路的COPS-SLS (Common Open Policy service-Service Level Specification)
來預留通道資源，以提升通話的 QoS ，實現了網際網路的通話品質是可以近乎目前的傳統網路(PSTN)的通話品質。再者利用網路電話架構之語音伺服器所求得之R Value作為分配語音流量之門檻值，使得雙方通話之中間轉送者語音伺服器具有穩定語音服務品質下進行分散負載流量之行為，此架構成為一個穩定的語音通話品質且實用性的網路電話架構。透過模擬結果顯示，本論文所提出的整合資源預留協定之SIP機制所呈現之單向封包延遲時間及單向封包遺失率，在實際的網路背景流量干擾下，效果優於未預留資源的狀態。語音品質監控門檻值則是利用RTCP封包之內容，取得必要之參數值後求得R Value，以R Value為主進行語音流量分散之演算，確定在穩定的通話服務品質下快速地分散網路電話之語音流量。

With the transmission of VoIP voice packets over the internet, maintaining consistent communication quality (Quality of Service, QoS) and efficient load balancing distributional structure is a mission critical.Since the SIP RFC 3602 standard does not articulate a specific protocol utilized to allocate dedicated broadband resources in different network environment for an user agent, this theses will addresses these inadequate point by proposing,considering the networking feature of SIP protocol and the VoIP applications, a integrated mechanism of the SIP reservation resource protocol to achieve consistent communication quality.Through the SIP mechanism of the integrated integrated-network RSVP(Reservation Service Protocol) and differentiated-network COPS-SLS (Common Open Policy service-Service Level Specification), these integrated mechanism is able to reserve VoIP channel resources, realizing the fact that the quality of the communication over VoIP is close to that over PSTN.Moreover, using the R Value calculated by the voice server under VoIP structure as a the threshold to allocate voice traffic brings about a steady communication quality for the voice packet relying server to practice the distributional load balancing act, a stable and practical VoIP structure.According to the simulation outcome, considering the delay time and the packet loss rate of one-way transmission under the practical interfered network environment, the result of adopting the SIP mechanism of the integrated reservation service protocol proposed in these theses is better than that of not adopting.The R Value, the monitoring threshold of the voice quality, is calculated by acquiring necessary parameters from the fields of RTCP packet. Furthermore, the use of the R Value to perform voice traffic distribution calculation is surely able to swiftly distribute VoIP traffic along with the steady communication quality.

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[17]QoS developed DiffServ functionality for NS-2 from http://ru6.cti.gr/ru6/.
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