| Summary: | 碩士 === 國立宜蘭大學 === 資訊工程研究所碩士班 === 100 === In the current stage of IPv6 (Internet Protocol version 6) VoIP (Voice over IP) deployment, an IPv6-enabled (i.e., IPv4/IPv6 dual-stack) VoIP phone may connect to an IPv4 VoIP phone or an IPv6 VoIP phone. When the IPv4/IPv6 dual-stack VoIP phone initiates a call by sending an IPv6 SIP (Session Initiation Protocol) INVITE message to an IPv4-only VoIP phone, the call cannot be established correctly. To resolve the problem, this paper investigates the server-based solutions such as SIP-ALG, redirect, and the CSCF-translation and this paper proposes an effective client-based solution, where the IPv4/IPv6 dual-stack VoIP phone is slightly modified to carry both IPv4 and IPv6 addresses in the SDP (Session Description Protocol) fields and the SIP server does not require any modification. The paper compares the proposed client-based solution with the server-based solutions in terms of network node modification, call setup complexity, and RTP (Real-time Transport Protocol) transmission latency. This paper also evaluates the PDD (Post Dialing Delay) and SIP server processing time. The results indicate that the proposed client-based solution outperforms the other three server-based solutions.
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