| Summary: | 碩士 === 國立東華大學 === 資訊工程學系 === 92 === Network technology make progress continually and popularity. Those factors make researchers interest in VoIP. Hope to transfer voice over IP networks and saving cost. H.323 (ITU-T) and SIP (IETF) are the two major approaches in VoIP. H.323 deploy hard and exist several drawbacks. So SIP by developing constructively. Since 1996 IETF joined into SIP’s research. RFC 2543, …RFC 3265 be announced gradually. SIP is an application-layer protocol. It is used to initial, modify, and terminate session. RFC 2543 defines six request methods. It includes Register, Invite, ACK, Option, Bye, Cancel, and defines six levels response code.
The thesis uses Client-server architecture. Server-end includes Proxy, Redirect and Registrar. Using web page to maintain user data, and database to manage sessions. Artiza is used to simulate SIP server and User Agent in real network. It can capture packet, analyze performance, packet lose and Quality of Service. Result of measured on end-end utilization is 0.17%, average initial time is 0.713785 seconds. End-Proxy-end utilization is 0.34%, average initial time is 1.443785 seconds. But initial time of H.323 in end-to-end is 0.875 seconds. So SIP has shorter initial time.
SIP has six characteristics. (1) SIP is an element protocol. H.323 is composed of H.225, H.245, H.246, H.332, H.450, H.26x, H.7xx. (2) SIP is a text-based, H.323 is a binary code. (3) SIP is easier to deploy and debug. (4) H.323 suits for LAN. Because it often routing loop. (5) SIP supports Call Hold, Call Forking and Third-Party Call Control. (6) H.323 only supports UDP and TCP protocol, and SIP can use any transport protocol. As a result, SIP still has space to develop
|
|---|
