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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 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.
Spirent SIP Server Platform; ShoreTel IP phone systems with unified communications and contact center built in; Snom One free/blue/yellow (Snom acquired and renamed pbxnsip) (SIP) Speedflow Communications VoIP class 4/5 softswitches with integrated billing, transcoding, SIP-H.323 converter.
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).
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 ...
The IETF produced a standard called the Session Initiation Protocol (SIP) that also enables voice and video communication over IP. There are also other ITU-T recommendations used for videoconferencing and videophone services – H.320 (using ISDN) and H.324 (using regular analog phone lines and 3G mobile phones).
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