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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]
As a simple example, at the edge of a corporate network, a firewall demarcates the local network (inside the corporation) from the rest of the Internet (outside the corporation). A more complex example is that of a large corporation where different departments have security needs for each location and perhaps for each kind of data.
An example of a SIP message exchange between two users, Alice and Bob, to establish and end a direct media session. SIP is only involved in the signaling operations of a media communication session and is primarily used to set up and terminate voice or video calls. SIP can be used to establish two-party or multiparty sessions. It also allows ...
Linux Netfilter's SIP conntrack helper fully understands SIP and can classify (for QOS) and NAT all related traffic; Netopia Netopia supports ALG; PF, built-in OpenBSD firewall PF can handle the NAT through the "static-port" directive and the bandwidth control through the built-in queuing system of SIP connections
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.
The architectural difference between SIP and H.323, and the media gateway control protocols is that the relationships between entities in SIP and H.323 are peer-to-peer, while the relationships between entities in media gateway control protocols use the master/slave (technology) model. SIP and H.323 handle call setup, connection, management ...
These two extensions allow users to specify their preferences about the service the IMS provides. With the caller preferences extension, [8] the calling party is able to indicate the kind of user agent they want to reach (e.g. whether it is fixed or mobile, a voicemail or a human, personal or for business, which services it is capable to provide, or which methods it supports) and how to search ...
SIP Application servers (AS) host and execute services, and interface with the S-CSCF using SIP. An example of an application server that is being developed in 3GPP is the Voice call continuity Function (VCC Server). Depending on the actual service, the AS can operate in SIP proxy mode, SIP UA mode or SIP B2BUA mode. An AS can be located in the ...