System Design Of Voice-Terminal For Network-conference Based On DSP

IP phone is a new technology about application of internet, which is aimed at transmitting real-time voice data over internet.Traditional IP phone must depend on PSTN or PC when working, so this thesis proposed a Design of Voice-Terminal for network-conference Based on DSP. This system consists of four subsystems: audio sample & convert, interface to network, information input & identify and the central processor of DSP. DSP chip is the substitute of traditional PC, whose main work is controlling both audio sample & convert subsystem and interface to network subsystem to realize AD/DA converting, packets transmitting and receiving over network, and that both speech-coding criterion and relevant internet protocols can also be integrated in DSP. Furthermore, with software improvement the functions of network-conference are expected: real-time voice data transmission between several Voice-Terminal. So the system is characterized on potable, easy to expanse, and independent of both PC and PSTN.Features and innovations compared to traditional IP phone are outlined firstly in thesis, then the principle of system working and important technologies about voice data transmission over internet are studied. Moreover the realization of Voice-Terminal scheme was researched including hardware & software design and debugging. We mainly discussed project design and subsystem division, DSP's resource distribution and communicating interface design between DSP and subsystems, chip selection and scheme design, printed circuit board(PCB) design and hardware's electric testing, subsystem's software debugging both in control and communication. Finally feasibility of the scheme is proved by system debugging including united of hardware and software.