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FreeSWITCH is a free and open-source telephony software for real-time communication protocols using audio, video, text and other forms of media. The software has applications in WebRTC , voice over Internet Protocol (VoIP), video transcoding, Multipoint Control Unit (MCU) functionality and supports Session Initiation Protocol (SIP) features.
This is a list of free and open-source software packages, computer software licensed under free software licenses and open-source licenses.Software that fits the Free Software Definition may be more appropriately called free software; the GNU project in particular objects to their works being referred to as open-source. [1]
JsSIP uses the SIP over WebSocket transport for sending and receiving SIP requests and responses, and thus, it requires a SIP proxy/server with WebSocket support. Currently the following SIP servers have been tested and are using JsSIP as the basis for their WebRTC Gateway functionality:
This is a comparison of voice over IP (VoIP) software used to conduct telephone-like voice conversations across Internet Protocol (IP) based networks. For residential markets, voice over IP phone service is often cheaper than traditional public switched telephone network (PSTN) service and can remove geographic restrictions to telephone numbers, e.g., have a PSTN phone number in a New York ...
Instead of forwarding call traffic through to an operator's mobile switching center, OpenBTS delivers calls via SIP to a VOIP soft switch (such as FreeSWITCH or yate) or PBX (such as Asterisk). This VOIP switch or PBX software can be installed on the same computer used to run OpenBTS itself, forming a self-contained cellular network in a single ...
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Codec2 has also been integrated into FreeSWITCH and there's a patch available for support in Asterisk. There was an FM-to-Codec2 digital voice repeater in earth orbit on amateur radio CubeSat LilacSat-1 (call sign ON02CN, QB50 constellation), which was launched and subsequently deployed from the International Space Station in 2017. [13]