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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]
SIP requests and responses may be generated by any SIP user agent; user agents are divided into clients (UACs), which initiate requests, and servers (UASes), which respond to them. [ 1 ] : §8 A single user agent may act as both UAC and UAS for different transactions: [ 1 ] : p26 for example, a SIP phone is a user agent that will be a UAC when ...
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.
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 ...
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.
FreePBX is a web-based open-source graphical user interface (GUI) that manages Asterisk, a voice over IP (VoIP) and telephony server. [2]FreePBX is licensed under the GNU General Public License version 3, [3] with commercial modules available under their own licenses.
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 ...
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.