Free minisip voip software for linux

Minisip project is a SIP User Agent ("Internet telephone") developed at KTH currently running on Linux. Keywords: Secure VoIP; SIP; MIKEY; RTP; SRTP; SDP; Video Telephony; Push-to-talk. You can download it for free from the download page.

Minisip is developed by Ph.D and Master students at the Royal Institute of Technology, KTH, Stockholm, Sweden.

The source code is available as a number of libraries under the GNU Lesser General Public License (LGPL) and applications under the GNU General Public Licence (GPL).

Here you will find tutorials, how-to guides, and instructions for setting up an enterprise network using free open-source software.

The primary goal is to show you how to setup your entire network using open-source solutions and save you a TON of money doing so.

And when we said entire network, we meant EVERYTHING including domain controlers, phone systems, email servers, file servers, backup servers, self-monitoring security systems...EVERYTHING!

We will provide guides that have as many screenshots, illustrations, or step-by-step "watch the movie" files as necessary to make the process as easy and clear as possible.
You can use the guides to complement your existing network or to replace existing components with more robust and less expensive solutions.

You will find that well step you through setting up a network that is enterprise grade. Well even show you how to inexpensively cluster your servers so downtime is something you dont even worry about anymore. And best of all the solutions will cost you practically nothing.

MD5: 6e098846a2c1265ba8b0a7d67ee7b5dc livecd.iso

PlayVoIP is a webbased VoIP service enabler featured with user management, statistical report and registration application system based on LAMP and Asterisk.

At the beginning this project was specially designed for simple and free Indonesian VoIP network, VoIP Rakyat.

Gratissip Tftp is a program that displays TCP/IP connections on an LCD display.
ratissip Tftpd is a TFTP server written in Java. It has special extensions which allows it to serve firmware and provisional settings for Grandstream VoIP phones.
Enhancements:
- The code was fixed up for announcement.

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.

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.

Siproxd is a proxy/masquerading daemon for the SIP protocol. It handles registrations of SIP clients on a private IP network and performs rewriting of the SIP message bodies to make SIP connections work via an masquerading firewall (NAT).
Siproxd project allows SIP software clients (like kphone, linphone) or SIP hardware clients (Voice over IP phones which are SIP-compatible, such as those from Cisco, Grandstream or Snom) to work behind an IP masquerading firewall or NAT router.
SIP (Session Initiation Protocol, RFC3261) is the protocol of choice for most VoIP (Voice over IP) phones to initiate communication. By itself, SIP does not work via masquerading firewalls as the transfered data contains IP addresses and port numbers.
There do exist other solutions to traverse NAT existing (like STUN, or SIP aware NAT routers), but such a solutions has its disadvantages or may not be applied to a given situation. Siproxd does not aim to be a replacement for these solutions, however in some situations siproxd may bring advantages.
HOW TO GET STARTED
make sure libosip2 is installed
If your libposip2 libraries are installed in /usr/local/lib, be sure to include this library path to /etc/ld.so.conf
$ ./configure
$ make
$ make install
edit /usr/etc/siproxd.conf according to your situation.
At least configure if_inbound and if_outbound. They must represent the interface names (e.g. on Linux: ppp0, eth1) for the inbound and outbound interface.
edit /usr/etc/siproxd_passwd.cfg if you enable client authentication in siproxd.conf
start siproxd (siproxd does not require root privileges)
$ siproxd
Enhancements:
- Several issues related to 64 bit architectures have been fixed and several minor bugfixes.