Title

具備錯誤回復和有效資源運用功能的語音電話系統

Translated Titles

Adopting SCTP and MPLS-TE Mechanisms in VoIP System for Fault Recovery and Resource Allocation

DOI

10.7117/JSET.201106.0077

Authors

薛來銘(Lai-Ming Shiue);張阜民(Fu-Min Chang);林志銘(Chih-Ming Lin);高勝助(Shang-Juh Kao)

Key Words

語音電話 ; 資源分配 ; 錯誤回復 ; 串流控制傳輸協定 ; 多重標籤交換協定 ; VoIP ; resource allocation ; fault recovery ; SCTP ; MPLS

PublicationName

科學與工程技術期刊

Volume or Term/Year and Month of Publication

7卷2期(2011 / 06 / 01)

Page #

77 - 87

Content Language

繁體中文

Chinese Abstract

網路語音電話的應用可以將語音資料在網路上傳輸。在網路語音電話的架構中,以會話發起協定(session initial protocol, SIP)做爲語音呼叫建立協定,並利用使用者資料協定(user datagram protocol, UDP)做爲SIP信令和語音封包的傳輸協定。因UDP協定並無擁塞控制的機制,若網路發生故障或壅塞,需藉由SIP應用程式的重傳機制來回復遺失的信令,此方式容易產生呼叫建立時間過長的問題,也無法提供語音封包傳輸品質的保證。因此,如何建立一個有效率及具有語音封包服務品質保證的網路語音電話架構,是一個值得研究的課題。在本文中,我們結合串流控制傳輸協定(stream control transmission protocol, SCTP)和多重標籤交換協定(multi-protocol label switching, MPLS)流量工程機制,提出一個具有錯誤回復和有效資源運用的語音電話架構。在所提出的架構中,SCTP協定被用來傳遞SIP信令,MPLS流量工程機制則被用來設立語音資料傳輸路徑。藉由SCTP協定多重位址的特性,可以降低語音呼叫失敗率及有效利用網路資源。透過MPLS流量工程機制的應用,網路鏈結或節點的中斷可以快速被恢復。我們使用網路模擬軟體(NS2)模擬三種不同架構並比較此三種架構在不同網路狀況時的效能。模擬結果顯示所提出的架構,語音呼叫的建立時間能維持在最佳的情況,語音封包傳輸的抖動率、封包漏失、和端對端延遲,都能達到語音傳輸服務品質的需求。

English Abstract

VoIP application allows the transmission of vocal data over Internet. In VoIP architecture, the Session Initiation protocol (SIP) is used in the setup of voice calls, while the User Datagram protocol (UDP) transports the SIP signaling messages and vocal data. Since UDP has no congestion control mechanism, the quality of vocal data transmission is not guaranteed and SIP must employ a retransmission mechanism for reliability, leading to possible performance degradations. Providing a reliable VoIP architecture that guarantees the quality of vocal data is an urgent research topic. This paper proposes a VoIP architecture capable of fault recovery and resource allocation by adopting the Multi-homed Stream Control Transmission protocol (SCTP) and Multi-Protocol Label Switching Traffic Engineering (MPLS-TE) method. In the proposed architecture, SCTP was employed to transmit SIP messages, while the MPLS-TE method was used to establish the transmitting path for vocal data. With the multi-homing capability of SCTP, the voice call failure rate can be reduced and network resources can be utilized efficiently. Through the MPLS-TE method, traffic engineering functions such as network resources optimization, strict quality of service (QoS), voice data delivery, and fast recovery upon link or node failure can be ensured. This study simulates three architectures using the network simulator (NS2) software and compares them under different network conditions. The simulation results demonstrate the applicability of the proposed architecture.

Topic Category 醫藥衛生 > 醫藥總論
醫藥衛生 > 基礎醫學
工程學 > 工程學綜合
社會科學 > 社會科學綜合
社會科學 > 心理學