An Implementation of SIP-based Analog Telephone Adapter on Embedded System

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 Pr...

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Main Authors: Ren-Jhe Wu, 吳仁哲
Other Authors: 高勝助
Format: Others
Language:zh-TW
Published: 2016
Online Access:http://ndltd.ncl.edu.tw/handle/20148292359523869609
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spelling ndltd-TW-104NCHU53940392017-01-11T04:08:09Z http://ndltd.ncl.edu.tw/handle/20148292359523869609 An Implementation of SIP-based Analog Telephone Adapter on Embedded System 在嵌入式系統上實作一個以SIP為基礎的網路電話適配器 Ren-Jhe Wu 吳仁哲 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. 高勝助 2016 學位論文 ; thesis 62 zh-TW
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description 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.
author2 高勝助
author_facet 高勝助
Ren-Jhe Wu
吳仁哲
author Ren-Jhe Wu
吳仁哲
spellingShingle Ren-Jhe Wu
吳仁哲
An Implementation of SIP-based Analog Telephone Adapter on Embedded System
author_sort Ren-Jhe Wu
title An Implementation of SIP-based Analog Telephone Adapter on Embedded System
title_short An Implementation of SIP-based Analog Telephone Adapter on Embedded System
title_full An Implementation of SIP-based Analog Telephone Adapter on Embedded System
title_fullStr An Implementation of SIP-based Analog Telephone Adapter on Embedded System
title_full_unstemmed An Implementation of SIP-based Analog Telephone Adapter on Embedded System
title_sort implementation of sip-based analog telephone adapter on embedded system
publishDate 2016
url http://ndltd.ncl.edu.tw/handle/20148292359523869609
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