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The Session Initiation Protocol ( SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. [1] SIP is used in Internet telephony, in private IP telephone systems, as well as mobile phone calling over LTE ( VoLTE ). [2]
Extended search being performed may take a significant time so a forking proxy must send a 100 Trying response. [1] : §21.1.1. 180 Ringing. Destination user agent received INVITE, and is alerting user of call. [1] : §21.1.2. 181 Call is Being Forwarded. Servers can optionally send this response to indicate a call is being forwarded. [1 ...
G.722 is an ITU-T standard 7 kHz wideband audio codec operating at 48, 56 and 64 kbit/s. It was approved by ITU-T in November 1988. Technology of the codec is based on sub-band ADPCM (SB-ADPCM). The corresponding narrow-band codec based on the same technology is G.726.
G.729. G.729 is a royalty-free [1] narrow-band vocoder -based audio data compression algorithm using a frame length of 10 milliseconds. It is officially described as Coding of speech at 8 kbit/s using code-excited linear prediction speech coding (CS-ACELP), and was introduced in 1996. [2]
SDES. SDES ( Session Description Protocol Security Descriptions) for Media Streams is a way to negotiate the key for Secure Real-time Transport Protocol. It has been proposed for standardization to the IETF in July 2006 (see RFC 4568 .)
The Session Initiation Protocol ( SIP) is the signaling protocol selected by the 3rd Generation Partnership Project (3GPP) [1] [2] to create and control multimedia sessions with multiple participants in the IP Multimedia Subsystem (IMS). It is therefore a key element in the IMS framework. SIP was developed by the Internet Engineering Task Force ...
The Session Description Protocol ( SDP) is a format for describing multimedia communication sessions for the purposes of announcement and invitation. [1] Its predominant use is in support of streaming media applications, such as voice over IP (VoIP) and video conferencing. SDP does not deliver any media streams itself but is used between ...
WebRTC Gateway connects between WebRTC and an established VoIP technology such as SIP. WebRTC (Web Real-Time Communication) is an API definition drafted by the World Wide Web Consortium (W3C) that supports browser -to-browser applications for voice calling, video chat, and messaging without the need of either internal or external plugins. [1]
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