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  2. Inter-Asterisk eXchange - Wikipedia

    en.wikipedia.org/wiki/Inter-Asterisk_eXchange

    IAX is a binary-encoded voice over Internet protocol (VoIP) that is used for streaming media, but is primarily designed for IP voice calls.. IAX uses a single User Datagram Protocol (UDP) data stream and port number, by default 4569, between endpoints for both session signaling and media payloads.

  3. User Datagram Protocol - Wikipedia

    en.wikipedia.org/wiki/User_Datagram_Protocol

    Most often, UDP applications do not employ reliability mechanisms and may even be hindered by them. Streaming media, real-time multiplayer games and voice over IP (VoIP) are examples of applications that often use UDP. In these particular applications, loss of packets is not usually a fatal problem.

  4. Voice over IP - Wikipedia

    en.wikipedia.org/wiki/Voice_over_IP

    In the European Union, the treatment of VoIP service providers is a decision for each national telecommunications regulator, which must use competition law to define relevant national markets and then determine whether any service provider on those national markets has "significant market power" (and so should be subject to certain obligations ...

  5. List of TCP and UDP port numbers - Wikipedia

    en.wikipedia.org/wiki/List_of_TCP_and_UDP_port...

    This is a list of TCP and UDP port numbers used by protocols for operation of network applications. The Transmission Control Protocol (TCP) and the User Datagram Protocol (UDP) only need one port for bidirectional traffic. TCP usually uses port numbers that match the services of the corresponding UDP implementations, if they exist, and vice versa.

  6. Session Initiation Protocol - Wikipedia

    en.wikipedia.org/wiki/Session_Initiation_Protocol

    The network elements that use the Session Initiation Protocol for communication are called SIP user agents. Each user agent (UA) performs the function of a user agent client (UAC) when it is requesting a service function, and that of a user agent server (UAS) when responding to a request. Thus, any two SIP endpoints may in principle operate ...

  7. Real-time Transport Protocol - Wikipedia

    en.wikipedia.org/wiki/Real-time_Transport_Protocol

    The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks.RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features.

  8. FreeSWITCH - Wikipedia

    en.wikipedia.org/wiki/FreeSWITCH

    FreeSWITCH is a free and open-source telephony software for real-time communication protocols using audio, video, text and other forms of media. The software has applications in WebRTC, voice over Internet Protocol (VoIP), video transcoding, Multipoint Control Unit (MCU) functionality and supports Session Initiation Protocol (SIP) features.

  9. List of SIP response codes - Wikipedia

    en.wikipedia.org/wiki/List_of_SIP_response_codes

    A specific flow to a user agent has failed, although other flows may succeed. This response is intended for use between proxy devices, and should not be seen by an endpoint (and if it is seen by one, should be treated as a 400 Bad Request response). [16]: §11.5 433 Anonymity Disallowed The request has been rejected because it was anonymous. [17]