多種編解碼器的網路電話語音品質量測

Abstract

本論文遵循 3rd Generation Partnership Project（3GPP）IP Multimedia Subsystem（IMS）系統所採用的 SIP（Session Initiation Protocol）、RTP（Real-time Transport Protocol）及 IPv6（Internet Protocol version 6）等通訊協定，在微軟 Windows Mobile 環境下開發一套網路電話軟體。此外，為了讓使用者能在通話前瞭解網路環境對語音品質的影響。本論文提出整合現狀服務量測機制，讓網路電話設備在通話前送出定量的封包，並迅速地評估出不同語音編解碼器的語音品質。使用者便能依照量測出的 MOS（Mean Opinion Score）值、封包遺失率（Packet Loss Rate）及延遲抖動（Jitter）等結果，決定是否要撥打電話。也能藉此選用通話品質較好的語音編解碼器，並以量測出的延遲抖動做為設定延遲抖動緩衝區的參考。

This thesis develops a Voice over IP (VoIP) software (i.e., VoIP phone) on Microsoft Windows Mobile system based on the protocols (e.g., Session Initiation Protocol, Real-time Transport Protocol and Internet Protocol version 6) adopted by 3rd Generation Partnership Project (3GPP) IP Multimedia Subsystem. For VoIP users, this thesis also integrates Instant Message and Presence Service (IMPS) system to develop a probing mechanism. In this mechanism, a VoIP phone sends the probing packets to a probing server before it dials up a call, and the probing server quickly calculates the Mean Opinion Score (MOS), packet loss rate and jitter based on the received packets. According to the MOS, the VoIP user can decide to dial a charged call immediately or later. The user also can select an appropriate codec and jitter buffer length based on the values of packet loss rate and jitter.