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<sflphoneteam@savoirfairelinux.com>
This program 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.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
Introduction
------------
SFLPhone is a Voice-over-IP software phone. We want it to be:
- user friendly (fast, sleek, easy to learn interface)
- corporate grade (transfers, holds, perfect audio quality)
- fully compatible with Asterisk (SIP and IAX protocols)
As the SIP/audio daemon and the user interface are separate processes,
it is easy to provide different user interfaces. SFLPhone0.8 comes with
a GTK graphical user interface, an interactive command line interface,
and even scripts to control the daemon from the shell.
SFLPhone is actually used by the support team of Savoir-Faire Linux Inc.
More information is available on the project homepage:
http://www.sflphone.org/
Getting the dependencies
------------------------
Required libraries and tools:
- ccRTP >= 1.3.4 (tested with ccrtp 1.4.1)
and Common C++ 2 >= 1.3.20 (tested with commoncpp2 1.4.1)
http://sourceforge.net/projects/gnutelephony/
- libeXosip2 >= 2.2.2 (tested with eXosip2 2.2.3)
and libosip2 >= 2.2.1 (tested with libosip2-2.2.2)
http://savannah.gnu.org/projects/exosip/
http://savannah.gnu.org/projects/osip/
- libxml2, libxml2-devel, pkgconfig
usually available in distribution repositories
Optional libraries:
- zeroconf: (tested with mDNSResponder87)
http://developer.apple.com/darwin/projects/bonjour/
- speex: codec for sound
Some of the required libraries may be downloaded from
http://www.sflphone.org/#downloads
or
http://www.sflphone.org/dev/
apt-get install autoconf automake1.9 libtool libosip2-dev libexosip2-dev libccrtp-dev libcommoncpp2-dev pkg-config libsamplerate-dev libasound2-dev
Building the dependencies
-------------------------
If you do not use either the development packages of your distribution or the source packages made by the upstream authors of dependencies, you may want to try our custom dependencies building script in tools/ directory:
Note that commoncpp, ccrtp, libosip and libexosip, samplerate are in debian and fedora.
You can also compile each dependency, one by one:
Note: if you install any package in /usr/local, don't forget to set pkg-config path with:
export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig/
If you have the sflphone 0.3 or 0.4 release, copy the file sflphonerc
to sflphonedrc:
cp ~/.sflphone/sflphonerc ~/.sflphone/sflphonedrc
If you have an older version, remove your sflphonerc file located in
$HOME/.sflphone
2. Lauch sflphone interface, a setup window appears.
3. If you do have a SIP account, fill the SIP stuff where related. You can
use STUN too.
4. Save your settings, restart it. You're done.
A user's guide will be available on:
http://www.sflphone.org/doc/
--------------------------
- Try using the same versions of the libraries as we do.
See http://www.sflphone.org in section Downloads.
- Try using the same compilers as ours: gcc 4.1.0 on Fedora Linux Core 5
is known to work.
- If you have dependencies installed in unusual directories, try setting
the environment variable CXXFLAGS before building, for example:
export CXXFLAGS="-I/opt/include"
Then, run ./configure with --enable-iax2 option.
Debugging SFLPhone
------------------
You can use the --with-debug option with configure
./configure --with-debug
make
Run-time troubleshooting
------------------------
- If errors occured when trying to load shared libraries, check that the
value of the environment variable LD_LIBRARY_PATH include the
directories containing the required libraries, eg:
export LD_LIBRARY_PATH=/path/to/my/lib:${LD_LIBRARY_PATH}
- Portaudio don't detect your sound card because :
* artsd is running
* jackd is running
On 2005/11/22, jack wasn't working with these settings:
jackd -d alsa -r 8000
* you use alsa with sflphone and another application, but dmixer isn't
configurated correctly inside ~/.asoundrc or /etc/alsa/alsa.conf
- sflphone refuse to start because it's running
You could have something like: "Exception: could not bind socket"
Try to close sflphone before
Short description of content of source tree
-------------------------------------------
- ringtones/ contains the different ringtones.
- stund/ is an implementation of the protocol STUN used when there is a NAT.
- utilspp/ allows to implement a singleton.
- src/ is the core of SFLphone. It contains the main.cpp, managerimpl.cpp
files, audio and gui directories, and files about signalisation SIP. Later,
it should be better, when IAX will be implemented, that a directory groups
these protocols.
The ManagerImpl class is the intermediaire between all the layer in the tree.
- src/audio/ is the audio layer. It contains all about tones, dtmf,
audiodriver, rtp layer, audio codec ulaw, alaw and gsm.
- src/audio/gsm/ contains the implementation of gsm audiocodec library.
- src/audio/pacpp/ implements PortAudioCpp, a native C++ binding of
PortAudio V19. (remove in sflphone 0.7)
- src/gui/ is the old directory that contains all about different user
interface.
- src/gui/server is the directory that talk (tcp socket on port 3999) to
sflphone client. The slfphone client can be a console program or a graphical
interface.
How is structured SFLphone (>=0.5)
----------------------------------
+------------+
|TCPSessionIO|
+------------+
|
+------------+
| SessionIO |
+------------+
|
+------------+
| GUI Server |
+------------+
|
+---------------------------+
| GUI Framework |
+---------------------------+
|
+---------------------------+
| ManagerImpl |
+---------------------------+
|
+---------------------------+
| Account |
+---------------------------+
|
+---------------------------+
| VoIPLink |
+---------------------------+
| |
+-----------+ +-------+
| SIP | | IAX |
| RTP | +-------+
| STUN |
+-----------+
It's in ManagerImpl where we declared Call vector and VoIPLink vector.
VoIPLink vector contains all different VoIP link like SIP, IAX, ...
Call vector contains all the calls according to the kind of VoIP link.
About Savoir-Faire Linux
------------------------
Savoir-Faire Linux is a consulting company based in Montreal, Quebec.
For more information, please check out our website:
http://www.savoirfairelinux.com/
Contributing to SFLPhone
------------------------
Of course we love patches. And contributions. And spring rolls.
The project infrastructure (CVS, bugtracker, MLs) is maintained on:
http://forge.novell.com/modules/xfmod/project/?sflphone
Do not hesitate to join us and post comments, suggestions, questions
and general feedback on the forge.novell mailing-list.