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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.
STUN is used in some of the sip phones to enable the SIP/RTP packets to cross boundaries of two different IP networks. A packet becomes unroutable between two sip elements if one of the networks uses private IP address range and other is in public IP address range. Stun is a mechanism to enable this border traversal.
A SIP provider (Session Initiation Protocol) is any telecommunications company which provides SIP trunking to customers, usually businesses. Many companies provide SIP "termination" (outbound calling) and "origination" (inbound calling, usually with a plain old telephone service (POTS) phone number, called a direct inward dialing (DID).
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]
Phone calls between subscribers of the same provider are usually free when flat-fee service is not available. [12] A VoIP phone is necessary to connect to a VoIP service provider. This can be implemented in several ways: Dedicated VoIP phones connect directly to the IP network using technologies such as wired Ethernet or Wi-Fi. These are ...
installing a SIP proxy, an XMPP/Jabber server, or both; installing IP phones, or softphone applications on computers or smartphones; To achieve maximum success, most deployments involve both SIP and Jabber, to ensure connectivity to any other party that supports only one or the other.
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