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C++ VoIP Software for BSD

Browse free open source C++ VoIP Software for BSD and projects below. Use the toggles on the left to filter open source C++ VoIP Software for BSD by OS, license, language, programming language, and project status.

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  • 1
    VoIP monitor

    VoIP monitor

    VoIP SIP and SKINNY quality analyzer and packet / audio recording tool

    VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY MGCP RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP / SKINNY protocol or SIP/RTP/RTCP/T.38/udptl protocols. VoIPmonitor can also decode audio.
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    Downloads: 765 This Week
    Last Update:
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  • 2
    Open Phone Abstraction Library (OPAL) is a C++ multi-platform, multi-protocol library for Fax, Video & Voice over IP and other networks. Also included is the Portable Tool Library (PTLib) which is a C++ multi-platform abstraction library and collection o
    Downloads: 72 This Week
    Last Update:
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  • 3
    Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.
    Downloads: 9 This Week
    Last Update:
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  • 4
    Stuntman - STUN server and client

    Stuntman - STUN server and client

    High performance, production quality STUN server and client library

    New version 1.2. This is the code to STUNTMAN - an open source STUN server and client code by john selbie. Compliant with the latest RFCs including 5389, 5769, and 5780. Also includes backwards compatibility for RFC 3489. ICE and WebRTC ready. Version 1.2 compiles on Linux, MacOS, BSD, and Solaris. Supports the STUN protocol on both UDP and TCP for both IPv4 and IPv6. Windows binaries are also provided. Additional features are in development. This is a mirror of the code on https://github.com/jselbie/stunserver More details on the project's website: http://www.stunprotocol.org
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    Downloads: 11 This Week
    Last Update:
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  • 5
    A command line SIP/H323 softphone capable of sending and receiving audio files as well as sending out of band DTMF digits. Supports a BNF format configuration language for scripting call scenarios. Useful for example system testing.
    Downloads: 2 This Week
    Last Update:
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  • 6
    GNU Gatekeeper (GnuGk)

    GNU Gatekeeper (GnuGk)

    H.323 Gatekeeper for VoIP and videconferencing

    The project has moved! Please find current versions at https://www.gnugk.org/ The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions. The project has moved! Please find current versions at https://www.gnugk.org/
    Downloads: 1 This Week
    Last Update:
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  • 7
    Taki

    Taki

    SIP softphone

    Cross-platform SIP softphone
    Downloads: 2 This Week
    Last Update:
    See Project
  • 8
    The goal of this project is to provide a liveCD allowing the user to analyze networks for VoIP installations. This project gives you a global network state.
    Downloads: 0 This Week
    Last Update:
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  • 9
    Fideliphone: High Quality Internet Telephony
    Downloads: 0 This Week
    Last Update:
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  • 10
    Open Unified Recording
    Open Unified Recording (OUR) is a full featured Linux based Open Source VoIP/SIP Call Recording engine, indexing and retrieval system. The system resides on the network and passively captures SIP sessions. Project is sponsored by UnifiedRecording.com
    Downloads: 0 This Week
    Last Update:
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  • 11

    OpenBTS

    GSM L1-L3 stack with SIP network interfaces

    OpenBTS is an implementation of the GSM air interface (Um) that allows cellular handsets to be used directly as SIP endpoints. It uses a software-defined radio to generate its air interface and uses Asterisk or yate as its network interface.
    Downloads: 0 This Week
    Last Update:
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  • 12

    SIP Anonymization Tool (SiAnTo)

    Small and effective program for SIP traces anonymization

    The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. A good way to design optimization techniques for SIP deployment would be to analyze SIP traffic from existing networks. However, publicly available analyses of SIP traffic are rare and thus not a lot of knowledge exists about typical behavior of a SIP server (as opposed to, for example, HTTP servers). Our aim is to promote analysis of real SIP-server behavior. Therefore, we present SiAnTo, an extended anonymization technique that substitutes session-participant information with matching, but nondescript, labels. This allows for SIP traces to be publicly shared, while keeping interesting traffic-session properties intact.
    Downloads: 0 This Week
    Last Update:
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  • 13

    SIP Data Filter (SiDaFir)

    Simple and efficient tool for SIP trace filtering

    The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. Thanks to its simplicity, SIP messages are often used in creative ways for which these were not originally designed (e.g. using periodical OPTIONS packets as NAT keep-alive instead of using STUN or TURN) and thus SIP traces of the captured traffic often contain "useless" traffic from the actual signalization point of view. Aim of this project is to offer a simple and effective, yet well-configurable, tool allowing for SIP trace filtering - the SIP Data Filter (SiDaFir).
    Downloads: 0 This Week
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  • 14
    SIP Express Media Server
    SEMS is a GPLv2+ media and application server for SIP based VoIP services. Basic applications like announcement, pre-call announcement, RBT, conference, voicemail, mailbox etc. and lots of example applications available.
    Downloads: 0 This Week
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  • 15
    Simu server
    Simu server project is aimed for IP telephony (VoIP) between classical SIP phones and Flash streaming applications.
    Downloads: 0 This Week
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  • 16
    VoIP Honey project provides a set of tools for building an entire honeynet, thus includes honeywall and honeypot emulating VoIP environments such as Asterisk PBX or OpenSer with fully configurable connections. Voip Honey runs on GNU/Linux and Windows Systems. It can be compiled for Mac OSX as well.
    Downloads: 0 This Week
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  • 17
    An XCAP server is used by XCAP clients to store data like buddy lists and presence policy in combination with a SIP Presence server that supports PUBLISH, SUBSCRIBE and NOTIFY methods to provide a complete SIP SIMPLE server solution.
    Downloads: 0 This Week
    Last Update:
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  • 18
    This is a SIP signaling layer to create a fully operative multipoint (video) conference server using SIP clients and RTP media streams in combination with strManager as a media management layer.
    Downloads: 0 This Week
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  • 19
    integral

    integral

    Fast CGI-based AsteriskPBX cluster administration tool.

    Allows OnLine monitoring of all kind Asterisk functioality: - Real-Time channel monitoring; - Queue Agents control; - Conference Administration; - Config edition; - Asterisk Core monotoring;
    Downloads: 0 This Week
    Last Update:
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  • 20
    nanoagi

    nanoagi

    Microscopic AGI framework.

    Asterisk Gateway Interface often used when it is necessary to build complex IVR system. Nanoagi will help to do it.
    Downloads: 0 This Week
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  • 21
    sippet

    sippet

    C++ SIP stack based on Chrome source code

    This C++ library has been designed as a Chrome SIP stack. Sippet is an open-source SIP User-Agent library, compliant with the IETF RFC 3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and P2P communication services. The main target was to enable Javascript applications to use UDP, TCP and TLS transports along WebSocket. Existing SIP solutions for the browser are forced to use the WebSockets API to send/receive SIP messages. But for many service providers, the WebSocket protocol could not be a feasible solution due to scalability, support and other business constraints. By the way, the stack can be used as a full client-side C++ Stack (full-featured) without Javascript integration.
    Downloads: 0 This Week
    Last Update:
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