Free conaito voip software for linux

Tapioca is a framework for Voice over IP (VoIP) and Instant Messaging (IM). Its main goal is to provide an easy way for developing and using VoIP and IM services in any kind of application.
Tapioca VoIP project was designed to be cross-platform, lightweight, thread-safe, having mobile devices and applications in mind.
Main features:
- Create a solution that integrates all components used by VoIP and IM applications in a single, reliable and easy to use framework, which is able to work on different platforms.
- Spare resources, providing central services for multiple applications. Eg.: The control of all incoming and outgoing SIP requests are managed by the SIP service, avoiding the creation of one SIP stack and allocation of a network port for each SIP-based application.
- Reduce the overhead of control layers and library dependencies.

Operator is a complete Linux (Debian) distribution that runs from a single bootable CD and runs entirely in RAM.
The Operator contains an extensive set of Open Source network security tools that can be used for monitoring and discovering networks.
This virtually can turn any PC into a network security pen-testing device without having to install any software. Operator also contains a set of computer forensic and data recovery tools that can be used to assist you in data retrieval on the local system.
Starting with the 3.3 version of Operator, we have started completely from scratch by installing a basic Debian installation then adding the KNOPPIX functionality afterwards. This allowed us to have more control and understanding of what is on the CD.
Main features:
- Debian based Linux Installation
- Linux-Kernel 2.4.31
- KDE V3.3.2-1
- wine Windows Emulator (Binary Emulator)
- Konqueror and Mozilla Firebird Web Browsers
- Koffice which includes korganizer, kword, kspread and more
- X Multimedia System (xmms) an MPEG-video, MP3
- Internet connection software kppp,pppoeconf (DSL)
- utilities for data recovery and system repairs, even for other operating systems
- network and security analysis tools for network administrators
- many programming languages, development tools
- in total more than 900 installed software packages with over 2000 executable user programs and utilities
- 100+ Unix/Windows Exploits and Tools ready to run
Enhancements:
- Modified wireless_select to use /proc/net/dev instead of /proc/net/wireless. Some cards were not showing up after they were reinserted like orinoco.
- Added package aim_1.5.286 AOL Instant Messenger
- Stripped down locales to use en_, de_, es_ only
- upgraded hydra-4.6 to hydra-4.7
- Added BusLogic driver to the kernel so that vmware would not panic when booting after an HD install.
- Updated Metasploit framework from 2.3 to 2.4
- reinstalled libnet1-dev
- fixed captive-ntfs
- Added new Exploits:
- HOD-ms05039-pnp-expl - (MS05-039) Microsoft Windows Plug-and-Play Service Remote Overflow
- HOD-kerio-firewall-DoS-expl - Kerio Personal Firewall Multiple IP Options Denial of Service
- HOD-ms04031-netdde-expl - Microsoft Windows NetDDE Remote Buffer Overflow Exploit
- HOD-ms04032-emf-expl - Microsoft Windows Metafile (.emf) Heap Overflow Exploit
- HOD-ms05002-ani-expl - Internet Explorer .ANI files handling Universal Exploit
- HOD-ms05017-msmq-expl - Message Queuing Buffer Overflow Universal Exploit
- DSR-cpanel - POC for Cpanel 5 and below
- cpanel-9x_RCE - POC for Cpanel 9 and below
- DSR-nethack - local exploit for Nethack 3.4.0
- phpLDAPadmin - phpLDAPadmin 0.9.6 - 0.9.7 Remote command Execution
- phpbb.php - phpBB 2.0.10 Remote command Execution
- HP_OV_NNM_RCE - HP OpenView Network Node Manager 6.2, 6.4, 7.01, 7.50 Remote Command Execution
- Added new Tools:
- zebra 0.94 - Tool that manages TCP/IP based routing protocols
- voipong 1.2 dev - VoIP call detector and voice dumper VoIPong is a utility which detects all Voice Over IP calls on a pipeline, and for those which are G711 encoded, dumps actual conversation to seperate wave files. It supports SIP, H323, Ciscos Skinny Client Protocol, RTP and RTCP.
- Upgraded yersinia v0.5.3 to v0.5.6 - Framework for performing layer 2 attacks
- ike-scan v1.2 - Discover and fingerprint IKE hosts (IPsec VPN Servers)

OSP Toolkit project is a client side implementation of the ETSI OSP VoIP Peering protocol (ETSI TS 101 321).

The OSP Toolkit project was begun in 1998 and the code has been incorporated into many commercial and open source VoIP products.

AdminsParadise VoIP PBX is a full Web-based phone and fax solution that integrates the best-of-breed open source VoIP software.
It runs Asterisk 1.4.1, hylafax, avantfax, and PHP5 with a themed, easy-to-use, Web-based interface.
Main features:
- Asterisk 1.4.1
- HylaFax
- AvantFax
- Scheduled Conferencing
- Integrated authentication
Enhancements:
- This initial release provides a live CD and an installation CD.

JVOIPLIB is an object-oriented Voice over IP (VoIP) library written in C++.
It is based upon work done for my thesis at the School for Knowledge Technology (or School voor Kennistechnologie in Dutch), a cooperation between the Hasselt University and the Maastricht University.
A part of this library was developed at the Expertise Centre for Digital Media (EDM) in Diepenbeek, Belgium. The EDM is a research institute of the Hasselt University.
Main features:
- Easy VoIP session creation and destruction.
- Highly configurable sessions: sampling rate, sample interval, compression type, ... can all be selected by the user. These features can also be changed during a session.
- Openness and extensibility: the object-oriented nature of the library makes it very easy to add features; new components can easily be tested by registering them as User Defined modules.
- Support for 3D effects: for my thesis I also did some research and development about VoIP in networked virtual environments, which included adding 3D effects to sound. For this reason, Ive added this feature to the library.

Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification.
Sofia-SIP project can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and person-to-person communication services.
The primary target platform for Sofia-SIP is GNU/Linux. Sofia-SIP is based on a SIP stack developed at the Nokia Research Center. Sofia-SIP is licensed under the LGPL.
Main features:
SIP features
- Sofia-SIP implementation follows RFC3261 and related key RFCs. INFO, UPDATE and REFER methods are supported. Also supported is SIMPLE presence and instant messaging, with the MESSAGE, SUBSCRIBE/NOTIFY and PUBLISH methods. Features such as early sessions, provisional responses, early media, caller preferences and session timers are included. Full set of transports, including both TCP and UDP over either IPv4 or IPv6, are supported.
SIP Offer-Answer module
- Sofia-SIP provides an implementation of the SDP offer-answer negotiation as specified in RFC3264. This is an essential component in using SIP to establish media sessions such as VoIP and video conferencing.
NAT traversal support
- Support for STUN as specified in RFC3489. STUN functionality is available via a separate module, so it can also be used independently from the base SIP stack. SIP extensions such as symmetric response routing (RFC3581/rport) are supported as well.
SIP security support
- Signaling can be secured by use of SSL/TLS. Also HTTP basic and digest authentication methods are supported.

Linux LiveCD VoIP Server can be used to provide a Vonage type service, or to create a voip pbx for a campus or business with up to thousands of SIP phones.
It is based on the Open Standard SIP Express Router (SER) and Asterisk. It can serve as a SIP Proxy, VoIP PBX, VoIP gateway or Class 5 Softswitch.
Main features:
- Easy Web user administration and real-time accounting.
- All in one solution to VoIP and SIP enable your business.
- Allows you to make your own SIP numbering plan. Centrex service.
- Can be connected to multiple A-Z wholesale termination providers and to your own PSTN termination gateway/router.
- Can do Least Cost Routing
- Includes nat traversal, stun server, media server for conference call bridge, voicemail to email, incomming virtual numbers (DIDs), follow me forwarding.
- Commercial pre-paid, post-paid and flat rate account support. No calling card (no b2bua in base system).
- Requires no software installation - it is a liveCD.
- Supports any SIP soft or hardware phones, such as popular XTen, Cisco ATA 186, Grandstream, Sipura, Bugetone, Linksys PAP2 and more.
- Supports SIP for Video Conferencing (Xten / CounterPath EyeBeam)
- Supports Ecrypted SIP with Xten / Counterpath Secure Xten Pro
- Requires a PC with fixed ip connection to the internet. 256 MBytes of RAM, CDRom reader, ide or sata hard disk to store call and user database and web site.
- Remote ssh configuration and administration help
Enhancements:
- Kernel 2.4.34.1 and minor bugfixes.