Open Source BSD Telephony Software

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.

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

Siproxd is a proxy/masquerading daemon for the SIP protocol. It allows SIP clients (softphones & hardphones) to work behind an IP masquerading firewall or router.

Gammu is a cellular manager for mobile phones/modems. It contains libraries and functions for ringtones,logos,phonebook,SMS,etc. (used by external software), a command line version (with backup/restore) and SMS gateway (with MySQL and PostgreSQL supp

Welcome to the Linux-IrDA project. The overall goal of this project is to make an implementation of the IrDA (tm) standards specifications for the Linux kernel. The code is licensed under the GNU Public licence (GPL) and is now included in Linux-2.2.

pcapsipdump is libpcap-based SIP sniffer with per-call sorting capabilities. It writes SIP/RTP sessions to disk in a same format, as "tcpdump -w", but one file per SIP session (even if there is thousands of concurrent SIP sessions). Getting started: http://pcapsipdump.sf.net/

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

Valhalla Chat is a Bluetooth Instant Messenger (IM), that provides a common messaging platform for mobile phones, desktop computers and pdas.

Peers is a very simple softphone. It's a SIP User-Agent, written in java, it works on windows, linux and mac. It can be used with SIP servers like opensips or asterisk IPBX. It supports G711 codec (PCMU and PCMA) and telephone-events (DTMF).

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

TurnServer is a implementation of Traversal Using Relay around NAT (TURN) protocol. This protocol allows a client to obtain IP addresses and ports from such a relay.

The libSRTP project has been moved to https://github.com/cisco/libsrtp

OpenGatekeeper is an Open Source H.323 Gatekeeper based on the work done by the OpenH323 project.

Plug and play provisioning server for snom VoIP phones written in Python. The server supports the plug and play provisioning feature embedded in snom phones of the snom 3xx, snom8xx and snom MeetingPoint series.

Git repo: https://github.com/asipto/siremis Web management interface for Kamailio (OpenSER) - handle subscriber profiles, access control lists, accounting, least cost routing and load balancing, monitoring charts, xmlrpc communication with SIP server

The ATSlog software provides a handy web-oriented interface for collecting, viewing and analysing calls for various types of PBX (Private Branch eXchange) models.

Cisco voice web interface is a web tool designed to help maintaining VoIP on cisco platforms. Based on a set of perl parser's it now allow's to perform quite all operation's needed to check voice call's. In close future planned to add dial-peer crea

Using the new C# Dll from swig portability of Openobex/ObexFtp to write a nice GUI able to transfer file between two devices with obex protocol. More Info’s here: http://cs-obexftp.wiki.sourceforge.net/

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.

Asterisco Settings is a Webmin module to configure and manage Asterisk PBX

AMS is a suite of software intended to make day to day administration and monitoring of an Asterisk PBX server easier. It contains a daemon that acts as a proxy to Asterisk's Manger Interface and a GTK GUI application for monitoring and administration.

Astmontray is a systray applet written in PyQt so users can monitor whether their PABX servers running Asterisk.org, the open source telephony platform, are up or not. It supports AMI authentication and runs fine on Linux, OSX and Maemo tablets.

My scripts I've made (or modified) to make my life easier.

A C++ framework utilizing Design Patterns for creating Linux and Windows communications applications that contain Dialogic® products. Includes media and network classes (analog, digital, SIP, H323), multithreaded event handling, distributed app support.