Remote Handset Voice Traffic

Apologies ahead of time but my Google skills appear to be failing me today but I just cannot seem to find a clear answer on this.

Let’s say we have two extensions (extensions 21 and 22 for instance) in a remote office that connects through the office’s site to site VPN tunnel back to the main office’s FreePBX box. If extension 21 wanted to call extension 22, how much of that traffic for the call would need to go over the VPN? My guess would be the SIP portions would have to go to the FreePBX box but the RTP traffic would go between the two phones.

Thanks in advance!

The answer is all of it. By default Asterisk (at least the FreePBX flavors) pass the audio for every call through the server, even if they are 2 phones on the same lan, with the same codec. There are ways to turn that off but my understanding is that it is still really picky and hard to make it actually hand off the audio.