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2. 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]

3. Session Initiation Protocol - Wikipedia

   en.wikipedia.org/wiki/Session_Initiation_Protocol

   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]

4. 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 ...

5. 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 ...

6. 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.

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: