Design Of SIP-based Embedded Voice Terminal

In the progress of the information age, people's demand for communication equipments is becoming more and more higher. They not only require them to have powerful funcition but also hope that its price is affordable and it can mobile flexibly. Taking account of the communication needs of end users, A SIP-based embedded voice terminal is designed and implemented in this paper selecting ADM5120 for hardware development platform. The terminal makes use of a wireless card which supports 802. llg to access the LAN and intelligently controls the entire conversation through the gateway application. It realizes the signaling interaction between terminal and terminal, terminal and server, and it also realizes the real-time transmission of media stream through RTP/RTCP. Basing on the deeply studying of the relevant protocols and key technologies, the design process of the terminal hardware and software is described in detail.In this paper, we first outline the research background and the research situation in and outside the country and propose the research object. Then we deeply analyze the basic function, architecture, message syntax and call flow of SIP and outlines the media negotiation protocol and IEEE802. llg standard involved in the communication process of SIP terminal. After that, we build the hardware platform following the overall design of the terminal's hardware architecture and discuss the main ADM5120 processor, VINETIC voice module and memory module and focuse on the software design part. The software design includes building the software platform and designing the software application. It firstly describes how to build cross-development environment. Then the wireless network kard driver and voice processing chip driver are introduced. After that, it focuses on the Linux operating system migration, including the migration of bootloader, kernel and root file system. In the application design part, we design the system into four modules and analyze the design process of its two major modules, SIP signaling module and gateway control module. The gateway application is designed depending on the logical relationship between each module. It generates the process of SIP user agent and RTP process. It also coordinates and controls other modules and is the manager of the entire software system. The core protocol stack is built into four layers based on the oSIP/eXosip protocol stack and achieves the function of UAC and UAS.Finally, making use of the desktop version of mini-SIP server and wireshark which is a kind of software to analyze network packets, we test the voice terminal from the aspects of interoperability, SIP signaling interaction, stability, anti-jamming and endurance and so on. Then the test results are analyzed. The results show that this terminal can correctly control SIP signaling and be in full compliance with RFC3261 standard. Its intercommunity is good and can support the intercommunity with traditional PSTN calls and SIP-Wifi phone. The quality of voice communication proves to be good.