The Function Upgrade Of Access Network Equipment Based On VoIP Technology

VoIP (Voice over IP) is the combination of telecom and computer technology, and it represents the development direction of future voice technology. If making comparison with the traditional PSTN network, the most significant advantage of VoIP is that it can utilize the present network resource at its widest extent and provide users with much more cheaper services. The technology in relation to VoIP is considerable and various, which are mainly developed into two directions. One is kept to the standard of ITU H.323. The other complies with the SIP protocol of IETF. In this paper, the author puts forwards a feasible project, which enables the traditional access network equipment to expand the IP functions through the VoIP technology.In the beginning chapter of this paper, the author presents the forming background and the developing history of VoIP, and then introduces the working principle, some key technologies, basic system structure and some standards that must be kept to of VoIP. Consequently, the author introduces the PSTN access network equipment and offers a complete project to expand the VoIP functions based on such equipment. By adopting H.323 as its design standard, concerning the features of access network, the author proposes the method to innovate the traditional VoIP gateway structure, based on which the VoIP system can be subdivided into three modules: gateway, gatekeeper and voice terminal.In the complete project, the author highlights the designs of the voice terminal module in VoIP and gateway module in H.323 respectively. During the process of voice terminal design, the author firstly presents its hardware structure, including the coding and encoding module of IP, the interface to former access network equipment and etc, and then the author illustrates in detail the software realization of G723.1 & G711 coder and encoder, then introduces how to realize the application of RTP/RTCP protocol in H.323 gateway by Openh323, including the design of the process and the realization of the data package format and partial code. The closing chapter is concentrated on the simulation of G723.1 &G711 coder and encoder. In this chapter, the coding and encoding effect of G723.1 is tested under different rates and voice sources.Though there still exist some problems in VoIP system for further exploration, the factors that constrain the functions of VoIP will be solved with the development of relative technology. It is sure that VoIP will replace the traditional voice-transmitting module so as to bring much more convenience to human being's daily life.