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A SIP phone is an IP phone that implements client and server functions of a SIP user agent and provides the traditional call functions of a telephone, such as dial, answer, reject, call hold, and call transfer. [16] [17] SIP phones may be implemented as a hardware device or as a softphone.
3Com VCX IP telephony module: back-to-back user agent SIP PBX; 3CX Phone System, for Windows, Debian 8 GNU/Linux; Aastra 5000, 800, MX-ONE; Alcatel-Lucent 5060 IP Call server; Aricent SIP UA stack, B2BUA, proxy, VoLTE/RCS Client; AskoziaPBX; Avaya Application Server 5300 (AS5300), JITC certified ASSIP VoIP; Bicom Systems IP PBX for telecoms
The term telephony application server [1] refers to an entity in a telephone network that carries out functions that are not directly related to the routing of messages through the network. Such functions can include in-network answering machines , 800 ( freephone ) numbers, automatic call forwarding , conference bridges and many other types of ...
The Session Initiation Protocol (SIP) is a signaling protocol used for controlling communication sessions such as Voice over IP telephone calls. SIP is based on request/response transactions, in a similar manner to the Hypertext Transfer Protocol (HTTP).
SIP trunking is a voice over Internet Protocol (VoIP) technology and streaming media service based on the Session Initiation Protocol (SIP) by which Internet telephony service providers (ITSPs) deliver telephone services and unified communications to customers equipped with SIP-based private branch exchange (IP-PBX) and unified communications facilities. [1]
A VoIP phone or IP phone uses voice over IP technologies for placing and transmitting telephone calls over an IP network, such as the Internet. [1] This is in contrast to a standard phone which uses the traditional public switched telephone network (PSTN).
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