| Summary: | 碩士 === 國立中興大學 === 資訊科學與工程學系 === 104 === In VoIP(Voice over IP) network environment connecting between PSTN(Public Switched Telephone Network) and VoIP is usually done through gateway or ATA(Analog Telephone Adapter) for voice signaling conversion. This function can be achieved by having IP PBX(IP Private Branch eXchange)with telephone card.
In this paper, we attempt to implement a cost-effective analog telephone adapter based on SIP via wired network, so you can call to a variety of SIP user agents through the conventional telephone system. Our development is based on CM5000 platform to complete the signaling conversions between SIP and PSTN. Provisioning of call features, such as call hold/retrieve, call forward, call transfer, call pick-up, fine me, music on hold, ring group, and voice compression and data conversion are accomplished through the underlying hardware implementation.
Of limited resources, this adapter can accomplish signaling conversion between SIP and PSTN, and also provides most common call features as supported in PSTN. The advantage of this device is convenient to apply in general residence or small enterprise with cost reduction as well as the provision of voice and data transmission.
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