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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.
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]
For media streams, the Real-time Transport Protocol (RTP) is used in most VoIP systems. For voice and media encoding, a variety of codecs are available, such as for audio: G.711, GSM, iLBC, Speex, G.729, G.722, G.722.2 (AMR-WB), other audio codecs, and for video H.263, H.263+, H.264. User interface software controls the operation of the ...
The prices of extensions on VoIP are lower than for PBX and key systems. VoIP switches may run on commodity hardware, such as personal computers. Rather than closed architectures, these devices rely on standard interfaces. [21] VoIP devices have simple, intuitive user interfaces, so users can often make simple system configuration changes.
Creacode SIP Application Server Real-time SIP call controller and IVR product for carrier-class VoIP networks; Dialogic Corporation Powermedia Media Servers, audio and video SIP IVR, media and conferencing servers for Enterprise and Carriers. Dialexia VoIP Softswitches, IP PBX for medium and enterprise organizations, billing servers.
The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks.RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features.
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Web-based VoIP is the integration of voice over IP technologies into the facilities and methodologies of the World-Wide Web. It enables digital communication sessions between Web users or between users of traditional telecommunication services.