Search Results for "c source code" - Page 3
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Showing 441 open source projects for "c source code"
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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... -
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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.
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atinout
AT commands as input are sent to modem and responses given as output.
This program will read a file (or stdin) containing a list of AT commands. Each command will be send to the modem, and all the response for the command will be output to file (or stdout). Example, to hang up any ongoing call: $ echo ATH | atinout - /dev/ttyACM0 - ATH OK $ -
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ATH
OSS code used in ATH products
Source code content for Open Source packages used by HMC220B, HMC3000 and HMC3021 products -
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USSDpad
Linux GUI USSD Sender
You can manage and send your ussd code with ussdpad .its simple and easy to use gui program written by use python and qt4 - Shashika Layan * Install Method (ArchLinux) It's Available on Arch User Repository (AUR) so you can install it use yaourt $ yaourt -S ussdpad -
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JAVIER
JAvascript Voicexml InterpretER
JAVIER is a JAvascript Voicexml InterpretER, designed (but not restricted) to run inside a web browser, its main engine has less than 1000 lines of code. It's maybe, the tiniest but (almost) FULL VoiceXML implementation. -
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This project implements a simple STUN server and client on Windows, Linux, and Solaris. The STUN protocol (Simple Traversal of UDP through NATs) is described in the IETF RFC 3489, available at http://www.ietf.org/rfc/rfc3489.txt
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DreamGallery
Source Code content for Open Source packages used by the DreamGallery
Source code content for Open Source packages used by the DreamGallery software. -
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usb-b2k
programm to handle a USB-B2K box
The program consist of an API (daemon) to handle a VOIP box of the type USB-B2K which allows to plug a classical telephone on an USB port (skypebox, wangobox, yealink...) in order to integrate this with any type of program of the type VOIP under GNU/LINUX. The functionalities are the following: 1. Switching to mode USB/PTSN 2. Detecting Pick up/ Hang up signal from the telephone 3. Generating Pick up tune 4. Generating Ring tone. 5. Handling telephone dial. This project is... -
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Video Content Management System
OSS code used in the Video Content Management System
Source code content for Open Source packages used by the Video Content Management System (VCMS) software. -
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CASMR GIL
Source Code content for Open Source packages used by the CASMR GIL
Source Code content for Open Source packages used by the CASMR Gateway Image Loader (GIL), which is a CASMR Plug-in module that allows the deployment of code download campaigns to the VMS and IP Client devices. -
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CASMR CDL
Source Code content for Open Source packages used by the CASMR CDL
Source code content for Open Source packages used by the CASMR Code Download (CDL). The code download portion of the Digicipher II system has been extended outside of the DAC 6000 and is now provided by CASMR 100. Code Download signaling or provisioning instructions are delivered through the out-of band control channel for DCII terminals. -
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CASMR DTA
Source code content for Open Source packages used by the CASMR DTA
Source code content for Open Source packages used by the CASMR DTA Plugin, which provides support for DTA devices. -
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CASMR SM
Source Code content for Open Source packages used by the CASMR SM
Source code content for Open Source packages used by the CASMR Service Management solution, which is a CASMR Plug-in module built on the CASMR Architecture. -
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CASMR CAST
Source Code content for Open Source packages used by the CASMR CAST
Source Code content for Open Source packages used by the CASMR Configuration Assistant (CAST). The CASMR Configuration Assistant will allow for the customer to easily organize the different versions of CASMR plugins and rapidly deploy software. This will allow the customer to create different areas (repositories) to distinguish plugins, such as ones that are in test vs. plugins that have been validated. -
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oSKY
Framework implementing SS7 over IP & SIGTRAN interworking and trunking
oSKY (OpenSKY) is a full communication framework to implement PSoTN (Public Social Telephony Network). In contrast with PSTN (Public Switched Telephony Network), oSKY intends to implement an open paradigm in the internet domain, gaining more than 100 years of experience from traditional telephone network. oSKY goals: -enhanced security (SSL, SCTP) -embedded NAT tunneling -Global title discovering on Internet style -Peer2Peer mode and Gateway mode -more... -
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mod_MumbleLink
Positional Audio Communication for Minecraft with Mumble
A Mod so that Minecraft now natively supports Mumble's positional audio feature. This means: Directional and positionally attenuated VOIP in relation to the game world. Please visit the Forum for information about the newest Version! Main Forum-Thread: http://www.minecraftforum.net/forums/mapping-and-modding/minecraft-mods/1272675 TheSkorm's Fork on GitHub: https://github.com/TheSkorm/mod_mumblelink Mumble: http://mumble.sourceforge.net Donations are greatly appreciated:... -
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GNU Telephony
GNU Telephony Services and Applications
GNU Telephony intends anyone to use free as in freedom software for telephony, and to do real-time collaboration freely and privately over the Internet, with the freedom to do so on any platform they choose to use. -
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2247-N8
Source Code content for Open Source packages used by the 2247-N8.
Source code content for Open Source packages used by the 2247-N8, which is a a DSL gateway CPE with wifi 802.11a/b/g/n and 4 port Ethernet switch. -
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Airean
GSM air interface
Airean is a proof-of-concept application showing that eavesdropping of GSM traffic (voice calls, mobile locations, SMS messages) is possible using a standard desktop PC with a small investment in hardware. The application requires a hardware interface to capture signals. Currently the following devices are supported: * Ettus Research USRP * Great Scott Gadgets HackRF (will be implemented when the device is released) Airean has the following capabilities: * Scan the GSM spectrum *... -
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PHP 2 Way Webcam Video Chat
1 on 1 Webcam Videochat Script with P2P Support
This is a web based instant 1 on 1 private online video conferencing solution. It's a solution for conducting easy to setup face to face meetings without leaving your office or home. It's the easiest and most cost-effective way to meet somebody and discuss one on one, to make a video call just by providing a private room access link. -
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Project moved: Find the latest code at https://www.h323plus.org/ Follows on from now depreciated OpenH323 and designed as a drop in replacement for OpenH323. Development of advanced open source H.323 including application sharing, video conferencing and incorporates new research and development work. Project moved: Find the latest code at https://www.h323plus.org/
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Voxy - the simplest way to integrate VoiceXML applications in Asterisk. Configure your dial plan with the URL of your VoiceXML application and it's done.
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baresip
Baresip is a modular SIP User-Agent with audio and video support
Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip
