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SIP is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP). [3] A call established with SIP may consist of multiple media streams, but no separate streams are required for applications, such as text messaging, that exchange data as payload in the SIP message.
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 intermediary machine or process rejected the call attempt. [25] This contrasts with the 607 (Unwanted) SIP response code in which a human, the called party, rejected the call. The intermediary rejecting the call should include a Call-Info header with "purpose" value "jwscard", with the jCard [26] with contact details. The calling party can ...
Furthermore, SBCs can actually make calls and other SIP scenarios work when they couldn't have before, by performing specific protocol "normalization" or "fix-up". In most cases, far-end or hosted NAT traversal can be done without SBCs if the VoIP phones support protocols like STUN, TURN, ICE, or Universal Plug and Play (UPnP).
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.
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