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MAD-FLUTE is an implementation of FLUTE - File Delivery over Unidirectional Transport (IETF draft). FLUTE is a protocol for the unidirectional delivery of files over the Internet, which is particularly suited to multicast networks. FLUTE builds on Asyncronous Layered Coding (ALC), the base protocol designed for massively scalable multicast distribution (RFC 3450). ALC is a protocol instantiation of Layered Coding Transport building block (LCT) (RFC 3451). MAD-ALC is an implementation of the ALC/LCT protocols. The MAD/TUT project is going on at Tampere University of Technology (TUT).

FindMTU is a tool that performs IPv6 path MTU discovery. You can use it to debug network problems and to detect IPv6-in-IPv4 tunnels in the path to a destination. FindMTU only performs IPv6 path MTU discovery. It does not know about IPv4.

FreeRADIUS is the most widely deployed RADIUS server in the world. It is the basis for multiple commercial offerings. It supplies the AAA needs of many Fortune-500 companies and Tier 1 ISPs. It is also widely used in the academic community, including eduroam. The server is fast, feature-rich, modular, and scalable.

FreeRADIUS is the most widely deployed RADIUS server in the world. It is the basis for multiple commercial offerings. It supplies the AAA needs of many Fortune-500 companies and Tier 1 ISPs. It is also widely used in the academic community, including eduroam. The server is fast, feature-rich, modular, and scalable.

AFS is a distributed filesystem product, pioneered at Carnegie Mellon University and supported and developed as a product by Transarc Corporation (now IBM Pittsburgh Labs). It offers a client-server architecture for federated file sharing and replicated read-only content distribution, providing location independence, scalability, security, and transparent migration capabilities. AFS is available for a broad range of heterogeneous systems including UNIX, Linux, MacOS X, and Microsoft Windows. IBM branched the source of the AFS product, and made a copy of the source available for community development and maintenance. They called the release OpenAFS.

FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many applications can be developed using a wide range of free tools.

It all started when we got some new routers, which told me the following when trying to upload configuration or download images from it: The TFTP server doesn't support the blocksize option. My curiousity was triggered, it took me some reading of RFCs and other documentation to find out what was possible and what could be done. Was plain TFTP very simple in its handshake, TFTP with options was kind of messy because of its backwards capability: The first packet returned could either be an acknowledgement of options, or the first data packet. Going through the source code of src/libexec/tftpd and going through the code of src/usr.bin/tftp showed that there was a lot of duplicate code, and the addition of options would only increase the amount of duplicate code. After all, both the client and the server can act as a sender and receiver. At the end, it ended up with a nearly complete rewrite of the tftp client and server. It has been tested against the following TFTP clients and servers: - Itself (yay!) - The standard FreeBSD tftp client and server - The Fedora Core 6 tftp client and server - Cisco router tftp client - Extreme Networks tftp client It supports the following RFCs: RFC1350 - THE TFTP PROTOCOL (REVISION 2) RFC2347 - TFTP Option Extension RFC2348 - TFTP Blocksize Option RFC2349 - TFTP Timeout Interval and Transfer Size Options RFC3617 - Uniform Resource Identifier (URI) Scheme and Applicability Statement for the Trivial File Transfer Protocol (TFTP) It supports the following unofficial TFTP Options as described at http://www.compuphase.com/tftp.htm: blksize2 - Block size restricted to powers of 2, excluding protocol headers rollover - Block counter roll-over (roll back to zero or to one) From the tftp program point of view the following things are changed: - New commands: "blocksize", "blocksize2", "rollover" and "options" - Development features: "debug" and "packetdrop" If you try this tftp/tftpd implementation, please let me know if it works (or doesn't work) and against which implementaion so I can get a list of confirmed working systems.

OpenLDAP is a suite of Lightweight Directory Access Protocol (v3) servers, clients, utilities and development tools. This package includes the following major components: * slapd - a stand-alone LDAP directory server * LDIF tools - data conversion tools for use with slapd This is the latest stable release of OpenLDAP Software for general use.

Mediastreamer2 is a powerful and lightweight streaming engine specialized in voice/video telephony applications. It is the library that is responsible for all the receiving and sending of multimedia streams in linphone, including voice/video capture, encoding and decoding, and rendering. Features: * Capture and playback from various platform dependent sound architectures * Send and receive RTP streams * Encode and decode the following audio formats: OPUS, speex, G711, GSM, iLBC, AMR, AMR-WB, G722, SILK, G729, and video formats H263, theora, MPEG4, H264 and VP8 * RTP/AVPF support: RTCP control messages for video error recovery: PLI, SLI, RPSI, FIR * Audio conferencing * Supports SRTP and ZRTP (encryption of voice and video) * Supports any webcam, based on OS's camera API: quicktime, directshow, video4linux, android.camera * Acoustic echo cancellation using the speex echo canceler or webrtc AECm * Read and write .wav files * Optimized rendering of YUV pictures, using OpenGL, DrawDib, X11/Xv * Dual tones generation * Custom tone detector * Audio parametric equalizer * Volume control, automatic gain control * ICE for optimized NAT traversal (RFC5246) to allow peer to peer audio and video connections without media relay server * Adaptive bitrate control algorithm: for automatic adaption of encoder bitrate based on received RTCP feedback * Can use plugins to add new codecs, new sound input/output backends,...

IP Messenger is a pop up style message communication software for multi platforms. It is based on TCP/IP(UDP). Win, Win16, Mac/MacOSX, X11R6/GTK/GNOME, Java, Div version and all source is open to public.