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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]
VoIP technology has evolved to integrate with mobile telephony, including Voice over LTE and Voice over 5G , enabling seamless voice communication over mobile data networks. These advancements have extended VoIP's role beyond its traditional use in internet-based applications.
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).
The U.S. National Institute of Standards and Technology (NIST), Advanced Networking Technologies Division provides a public-domain Java implementation [28] that serves as a reference implementation for the standard. The implementation can work in proxy server or user agent scenarios and has been used in numerous commercial and research projects.
In enterprise environments, VoIP technology enables the consolidation of voice and data networks into a unified IP infrastructure. This consolidation eliminates the need for separate voice and data pipelines while providing advanced capabilities including: Unified communications integration; Presence management systems; Advanced call routing
Mobile VoIP client: Consumers have started to use telephony interfaces on their PCs. Applications offer a low-cost, convenient way to access telephony services while traveling. Now mobile operators can offer a similar service with a UMA-enabled mobile VoIP client. The client provides a mirror interface to a subscriber's existing mobile service.
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