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  2. 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:

  3. List of SIP software - Wikipedia

    en.wikipedia.org/wiki/List_of_SIP_software

    A SIP server, also known as a SIP proxy, manages all SIP calls within a network and takes responsibility for receiving requests from user agents for the purpose of placing and terminating calls. Asterisk; ejabberd; FreeSWITCH; GNU SIP Witch; Kamailio, formerly OpenSER [1]

  4. 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.

  5. List of SIP response codes - Wikipedia

    en.wikipedia.org/wiki/List_of_SIP_response_codes

    [1]: §21.6.1 603 Decline The destination does not wish to participate in the call, or cannot do so, and additionally the destination knows there are no alternative destinations (such as a voicemail server) willing to accept the call. [1]: §21.6.2 The response may indicate a better time to call in the Retry-After header field.

  6. Packet analyzer - Wikipedia

    en.wikipedia.org/wiki/Packet_analyzer

    Documenting regulatory compliance through logging all perimeter and endpoint traffic; Gain information for effecting a network intrusion; Identify data collection and sharing of software such as operating systems (for strengthening privacy, control and security) Aid in gathering information to isolate exploited systems; Monitor WAN bandwidth ...

  7. STUN - Wikipedia

    en.wikipedia.org/wiki/STUN

    STUN is a tool used by other protocols, such as Interactive Connectivity Establishment (ICE), the Session Initiation Protocol (SIP), and WebRTC. It provides a tool for hosts to discover the presence of a network address translator, and to discover the mapped, usually public, Internet Protocol (IP) address and port number that the NAT has ...

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