The codecs are loaded at the initialization of the rtp session and destroyed at the end.
GSM audio quality is far better now. Because the dynamic library had to load at every 20ms the libgsm, it introduced a delay at each frame

!!!!!!!!!!!  Must have libspeex1 and libspeex1-dev installed

The previous commit brought a segmentation fault when launching the client.
When the daemon is lauched separatly, it uses a wrapper script, but it doesn't when the client start the daemon.
SO the LD_LIBRARY_PATH was ignored and the daemon couldn't find the dynamic library for the codecs.

Codecs dynamic libraries are created at the compilation time if they don't exist and copied in /usr/lib/sflphone/codecs at the installation

Audio calls with ilbc codecs still not tested.

Note: Manually copy the .so library in /usr/lib , then ldconfig

Copier les librairies dynamiques (codec_*.so) dans le répertoire /usr/lib et upda
faire ldconfig

Sampling rate values are no more hardcoded. The sampling rate of the audio layer
 and the frame size can be set in the user config file. The clock rate of the codec we use
in the rtp session is set with his actual value, but can be changed by modifying
the available codec in the user config file (only G711 for now)

Tiny doc fixes and typos.

This was due to libsamplerate's call to src_simple, which created glitches.
Now we use src_process, the 'Full API' as written on the SRC (libsamplerate)
website: http://www.mega-nerd.com/SRC/api_full.html

+ Add .gitignore

Samplerate is now a dependancy.

(adding codec map to new incoming call)

Adding debugging information

Fix onhold/getcallstatus issues

New account implementation.
Need to test more and add dynamic handling in Qt Interface.
SipCall, Call, VoIPLink were rewritten completly
Configuration now in [SIP0] section of .sflphonedrc
The debug of sflphoned is very verbose.
sflphone-cli may not work right now (need to send SIP0 as accountId).

VoIPLink integration inside account
getAccount() patch for managerimpl
Remove old voiplink vector in managerimpl

New STUN logic to discover true firewall port (NAT)
Send true IP/port in SDP message
Remove big ###### debugging message
New logic to listening on SIP port: 2 tries

Remove debug message
Add changes to project files
Fix a nonat-nonat bug (reinvite)

Fixe issue for current call id switching and for incoming call signal

Only switch call when no one are selected

ReInvite implementation for two no-nat telephone

Handling onhold answering call

Don't close sflphone when portaudio failed to load
Remove enable_audio variable (non-protected) in SipCall