tSIP: RTP streaming

Example setup with button starting RTP streaming and few buttons switching audio source to audio file, radio station or microphone.

Notes

wave files must be mono
for wave files: streaming ends when file ends
streaming target (multicast or unicast address with port) is specified as "Number" from the button; multiple buttons for different targets can be defined
wave file can be assigned directly to each button starting RTP streaming, allowing to play announcements with single click
G.722 is selected (in RTP button configuration) in this example; L16/44100/10 would have better audio quality, but most of regular phones would not be able to receive it
apparently some crappy phones are accepting only multicast (not unicast)
ffmpeg avformat behaves in a different way than regular wav player, repeating file instead of disconnecting - for consistency converting everything to wav might be recommended

Complete binary set: rtp_streaming.7z.