| Summary: | 碩士 === 國立中山大學 === 資訊工程學系研究所 === 94 === As network bandwidth growing and voice coding enhancing, voice transmitted over the Packet-Based network environment can also have good quality compared to traditional telephone network. Therefore, the IP telephony services having a price advantage could gradually replace the traditional telecommunication services. However, in order to support both the traditional telecommunication services and IP phone services, it requires the devices which are capable of converting analogical telephony information such as voice and fax into packet data suitable for transmitting over IP. We therefore design and implement an IP phone equipment that can make SIP phone calls, and can support the exchange of the analogical traditional telephone voice and the digital voice packets over IP.
In the hardware design point of view, we implement an IP phone set device in an embedded development platform using Infineon EASY5120 development kit, which uses a digital signal processor, called Vinetic®-2CPE (part number is PEB3332), to handle voice encoding and decoding, e.g., G.711 u/A law and G.723 compression, and RTP encapsulation and decapsulation. In the software architecture, we choose Linux as our embedded operation system under which there are lots of GNU open source software to feed our need and further to develop our own software components. In the process of this implementation, the software and hardware co-design takes up most of time, and we also face some VoIP application problems, e.g., SIP. We try to build up a VoIP system to figure out and solve these problems. And we hope that this actually applicable VoIP embedded system can be used as a testbed and verifying platform for VoIP applications.
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