應用於以SIP為基礎的VoIP環境之IP位址轉換機制

Abstract

在以Session Initiation Protocol(SIP)為基礎之Voice over IP (VoIP)服務中，網路電話使用IP位址表示其所在位置，因此IPv4位址不足的問題會影響SIP多媒體服務的佈建。目前常見的解決方法是，在IPv4網路中佈建網路位址轉換器(Network Address Translator; NAT)，使得多支網路電話共用一個IPv4位址，但如此卻導致SIP/Real-time Transport Protocol (RTP)穿越NAT的問題發生。為此我們研發出支援現有解決穿越NAT問題方法的網路電話軟體，並討論各種解決方法的優缺點。而另一解決IPv4位址不足問題的方法是採用IPv6來提供足夠的位址，使得每支網路電話均有其獨一無二的IPv6位址。只是在佈建IPv6初期，必須面臨IPv4與IPv6互通而產生的問題。對於此問題，我們提出一有效率的解決方法，其在通話建立以及語音傳送的效能表現皆優於現有方法。根據我們的分析，穿越NAT的問題尚未有十全十美的解決方法，而使用IPv6搭配我們所提出解決IPv4與IPv6互通問題的方法，對於佈建以SIP為基礎之VoIP服務是最有效率的方案。

In Session Initiation Protocol (SIP)-based Voice over IP (VoIP) services, IP addresses are used as location information of User Agents (UAs). The insufficiency of IPv4 addresses becomes a problem in SIP-based VoIP deployment. Deploying Network Address Translators (NATs) into IPv4 networks is one approach so that a public IPv4 address is shared by multiple UAs. However, this approach leads to SIP/Real-time Transport Protocol (RTP) NAT traversing problem. We design and develop a UA that supports the existing SIP/RTP NAT traversing solutions. This UA is used for further analyses and experiments. Another approach is to adopt IPv6 so that every UA owns a unique IPv6 address. Nevertheless, IPv4/IPv6 interworking problem for SIP-based VoIP may occur in the early stage of IPv6 deployment. To solve this problem, we propose an effective solution where the least call setup overhead is introduced and no impairment is incurred to the RTP transmission performance. Our study indicates that none of the existing SIP/RTP NAT traversing solutions is perfect in all aspects. Instead, the usage of IPv6 cooperating with our proposed solution is the best choice with the most efficient performance for deploying SIP-based VoIP services.