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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]
For example, if user A is talking to user B via a proxy P, SDES allows negotiation of keys between A and P or between B and P, but not between A and B. For end-to-end media security you must first establish a trust relationship with the other side.
Reliability is achieved by the sender detecting lost data and retransmitting it. TCP uses two primary techniques to identify loss. Retransmission timeout (RTO) and duplicate cumulative acknowledgments (DupAcks). When a TCP segment is retransmitted, it retains the same sequence number as the original delivery attempt.
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 ...
H.323. H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. [1] The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point ...
G.722 is an ITU standard codec that provides 7 kHz wideband audio at data rates from 48, 56 and 64 kbit/s. This is useful for voice over IP applications, such as on a local area network where network bandwidth is readily available, and offers a significant improvement in speech quality over older narrowband codecs such as G.711 , without an ...
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]
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 ...