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  2. Session Initiation Protocol - Wikipedia

    en.wikipedia.org/wiki/Session_Initiation_Protocol

    SIP user agent registration to SIP registrar with authentication. A registrar is a SIP endpoint that provides a location service. It accepts REGISTER requests, recording the address and other parameters from the user agent. For subsequent requests, it provides an essential means to locate possible communication peers on the network.

  3. List of SIP response codes - Wikipedia

    en.wikipedia.org/wiki/List_of_SIP_response_codes

    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 ...

  4. STUN - Wikipedia

    en.wikipedia.org/wiki/STUN

    STUN (Session Traversal Utilities for NAT; originally Simple Traversal of User Datagram Protocol (UDP) through Network Address Translators) is a standardized set of methods, including a network protocol, for traversal of network address translator (NAT) gateways in applications of real-time voice, video, messaging, and other interactive communications.

  5. SIP trunking - Wikipedia

    en.wikipedia.org/wiki/SIP_trunking

    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]

  6. SIP URI scheme - Wikipedia

    en.wikipedia.org/wiki/SIP_URI_scheme

    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 ...

  7. JsSIP - Wikipedia

    en.wikipedia.org/wiki/JsSIP

    JsSIP uses the SIP over WebSocket transport for sending and receiving SIP requests and responses, and thus, it requires a SIP proxy/server with WebSocket support. Currently the following SIP servers have been tested and are using JsSIP as the basis for their WebRTC Gateway functionality:

  8. Category:AD 6 endings - Wikipedia

    en.wikipedia.org/wiki/Category:AD_6_endings

    Main page; Contents; Current events; Random article; About Wikipedia; Contact us; Pages for logged out editors learn more

  9. Category:AD 6 - Wikipedia

    en.wikipedia.org/wiki/Category:AD_6

    Main page; Contents; Current events; Random article; About Wikipedia; Contact us; Help; Learn to edit; Community portal; Recent changes; Upload file