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This response may be used to send extra information for a call which is still being set up. [1]: §21.1.5 199 Early Dialog Terminated Can be used by User Agent Server to indicate to upstream SIP entities (including the User Agent Client (UAC)) that an early dialog has been terminated. [3]
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]
Asterisk is a software implementation of a private branch exchange (PBX). In conjunction with suitable telephony hardware interfaces and network applications, Asterisk is used to establish and control telephone calls between telecommunication endpoints such as customary telephone sets, destinations on the public switched telephone network (PSTN) and devices or services on voice over Internet ...
SIP works in conjunction with several other protocols that specify the media format and coding and that carry the media once the call is set up. For call setup, the body of a SIP message contains a Session Description Protocol (SDP) data unit, which specifies the media format, codec and media communication protocol.
A session border controller (SBC) is a network element deployed to protect SIP based voice over Internet Protocol (VoIP) networks. [1]Early deployments of SBCs were focused on the borders between two service provider networks in a peering environment.
The SIP URI scheme is a Uniform Resource Identifier (URI) scheme for the Session Initiation Protocol (SIP) multimedia communications protocol. A SIP address is a URI that addresses a specific telephone extension on a voice over IP system. Such a number could be a private branch exchange or an E.164 telephone number dialled through a specific ...
Based on these inherent SIP features it is possible to construct a peer-to-peer network of SIP nodes. In another approach, SIP over P2P, the SIP location service is replaced by conventional P2P overlay networking approach, such as the OverSim framework. [5] In this model, the overlay network is used for service or node discovery and rendezvous.
Inter-Asterisk eXchange (IAX) is a communications protocol native to the Asterisk private branch exchange (PBX) software, and is supported by a few other softswitches, PBX systems, and softphones. It is used for transporting voice over IP telephony sessions between servers and to terminal devices.