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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.

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/

This ASN.1 compiler turns ASN.1 specifications into C code. The asn1c is shipped together with conformant BER/DER/XER/PER codecs. The X.509, GSM TAP3, MEGACO, RRC and LDAP encoding and decoding examples are part of the source code distribution. NOTE: THE asn1c PROJECT HAS LARGELY MOVED TO GITHUB: http://github.com/vlm/asn1c

This project will allow your applications to send or receive GSM messages.

The OpenSIPStack Library is an implementation of the Session Initiation Protocol as described in RFC 3261. Applications: * OpenSBC (B2BUA) * OSSPhone (Softphone) Source code is available at http://www.opensipstack.org.

*** MOVED ON GOOGLE CODE ***

Java version of Dialogic's SwitchKit API. This API is actually a more "Java Centric" API, with listeners, filters, and a very extensible and modular framework that finally uses JNI to access the native libraries provided by Dialogic.

BladeWareVXML is a portable VoiceXML 2.1 interpreter that is an enhanced version (performance, usability and integration) of OpenVXI. A commercial version, with documentation, sample code, and support options, is available from the Commetrex Website.

Implementation of Media Resource Control Protocol Client (MRCP). Supports ASR and TTS functionality. Design pattern implementation. Documentation, sample application and library source code.

DialogPalette is a visual, integrated design environment for the creation of telephony applications running on the Asterisk Open Source PBX.

OSG ( OpenH323 Study Group ) has many sample programs (very basic, code base) which would help you to understand how OpenH323 works. The mailing list of OSG would be a good place to ask about basic information OpenH323.

Time of day service over telephone using Voicent Gateway, a VoiceXML gateway that specially designed for voice modems. A Free version is available for download at http://www.voicent.com/download. Sample code for interactive telephony applications.

An open project for a number of activex controls for use in the Windows environment. The controls have been sold commerical before by Eurosource (http://www.eurosource.se) but are now free complete with source code.

The purpose of this project is creating a plugin for <a href="http://www.ethereal.com/">Ethereal</a> that will enable it to decode ITU-T Recommendation H.323 traffic. The implementation is based on <a href="http://www.openh323.org">OpenH323</a> code

An Open H.323 based channel driver for the the Asterisk PBX. This code has now been intergrated into the Asterisk source tree. Please see the Asterisk source tree for any future updates of this software.

PNX system develops the middle-ware source code to glue Asterisk with a number of powerful telephony products such as: 1) OpenH323 H.323 stack 2) Vovida SIP stack 3) Bayonne Voice Automation Platform The advantages? An advanced IP PBX supporting the wide

This is a Win32, VC++ MFC application to manage Nokia mobile phones melodies. Includes VC++ source code and Rational Rose UML model.

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...