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A SIP provider (Session Initiation Protocol) is any telecommunications company which provides SIP trunking to customers, usually businesses. Many companies provide SIP "termination" (outbound calling) and "origination" (inbound calling, usually with a plain old telephone service (POTS) phone number, called a direct inward dialing (DID).
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]
An Internet telephony service provider (ITSP) offers digital telecommunications services based on Voice over Internet Protocol (VoIP) that are provisioned via the Internet. ITSPs provide services to end-users directly or as whole-sale suppliers to other ITSPs.
Both signaling (such as SIP) and media (such as RTP) is transported end-to-end in accordance with IPX specifications. A typical end-to-end path of traffic in a fixed to mobile interconnection scenario is illustrated below: X is a peering point where IPX Provider A and IPX Provider B exchange traffic
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 . [2]
Voxbone S.A. is a global communication as a service (CaaS) company that is a wholly owned subsidiary of Bandwidth, Inc., with offices in locations including Brussels, London, San Francisco, Austin, Simi Valley, Dublin, Singapore and Iași. Voxbone became a part of Bandwidth on November 2, 2020.
Aricent SIP UA stack, B2BUA, proxy, VoLTE/RCS Client; AskoziaPBX; Avaya Application Server 5300 (AS5300), JITC certified ASSIP VoIP; Bicom Systems IP PBX for telecoms; Brekeke PBX, SIP PBX for service providers and enterprises; Cisco SIP Proxy Server, Cisco unified border element (CUBE), Cisco Unified Communication Manager (CUCM)
Kamailio, formerly OpenSER (and sharing some common history with SIP Express Router (SER)), is an SIP server licensed under the GPL-2.0-or-later license. It can be configured to act as a SIP registrar, proxy or redirect server, and features presence support, RADIUS / syslog accounting and authorization, XML-RPC and JSON-RPC-based remote control, SQL and NoSQL backends, IMS / VoLTE extensions ...
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