Sorry if this is a noob question, but is there any way to just use raw pcm between two SIP clients? It seems super hard to find out what the very best audio quality is that you can use in phone systems in general. i.e. Is it possible to just do 16/44 or 24/96 LPCM between devices instead of all these weird codecs? I’m talking about devices on a LAN, not going out over the internet.
Also, if it is possible, what RTP devices and/or software supports this??
Thanks in advance,