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SICSOPHONE is a real-time, IP-based system for point-to-point delivery of audio between computer end-systems.

Sicsophone is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version.

The aim of the sicsophone is to provide a free and open telephony platform. The name sicsophone is of historical reasons. It wasn't really developed at SICS. We just didnt find a better name than that at the time of writing a paper. Maybe we should call it fiveophone?

The directory contains the following files and directories:

audio Audio-interface specific modules coding Audio-encoding modules configure File generated from configure.in with autoconf configure.in Meta file for autoconf configuration COPYING GPL 2 license docs Documentation. There is a useful architecture description. include Include files src Source code targets Sub-directory containing binaries for several target archs. README This file

The code is more or less modularized. More in the parts where it should be extensible. Less in the parts that are not expected to change. The extensible parts are audio and coding. These two sub-directories contain independent modules that implement different audio interfaces (eg /dev/audio, MS DirectSound, file emulation, etc); and coding algorithms (eg PCM, G711 ulaw and mlaw...).

The extensible modules follow a pseudo object-oriented methodology and it should be enough to change in these directories when you wish to add a new module. You may also have to change in the user interfaces (for obvious reasons). The audio and coding directories contain README's on how to add a new module. The architecture description in the docs directory may also be useful.

Otherwise, the code follows two basic tracks illustrated by the two example applications: sphone_main_send and sphone_main_rcv. The first (sphone_main_send) records audio samples, encodes them and encapsulates them and sends them on an RTP/UDP/IP socket. The second (sphone_main_rcv) receives RTP/UDP/IP packets, decodes them, arranges the samples according to a playout algorithm (fifo or shbuf) and then plays them.

There is currently no (1) signaling (2) GUI. Please help.

Other code weaknesses can be seen in the docs/todo.txt file.

To get going:

1. cd to target directory: Example:

cd targets/win32

2. Execute configure with correct parameters (check with configure --help Example:

../../configure

3. Compile sources into executables:

make

4. Run the executables:

cd app sphone_main_rcv & sphone_main_send &