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

Samplerate is now a dependancy.

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).

ReInvite implementation for two no-nat telephone

Remove some comment
Correcting some sound issue
Using fix FRAME_PER_BUFFER is better than 0

Now, only the audiolayer control the volume

Cleaning compilation and debug message

Add my copyright to the file
Add a CodecMap (the map) and CodecDescriptorMap (the one that initialize it)
for the request: list seq1 codecdescriptor

Fix volume control

Add notification incoming call.

Change setup directory name to ".sflphone".
Fix volume control.