I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
Moved public API classes into namespace tgvoip (CVoIPController -> tgvoip::VoIPController, CVoIPServerConfig -> tgvoip::ServerConfig)
Endpoint is now a class instead of a struct; also, IP addresses are now wrapped into objects instead of relying on in_addr and in6_addr
Full Windows port (Win32 threading + Winsock + WaveOut/WaveIn)
Added support for ALSA audio I/O on Linux (closes#2)
Abstracted away low-level networking to make it more portable
Minor bugfixes
