Audio session route

Let i say for example i set up an asterisk with 2 sip phones and 1 sip trunk.
sipphone 1 ip: 192.168.1.100
sipphone 2 ip: 192.168.1.101
trunk ip: 194.144.23.45
asterisk ip : 192.168.1.50

I wonder which route the audio stream takes.

From phone 1 to phone 2 will the audio stream be routed through the asterisk ( audio stream goes 192.168.1.100 -> 192.168.1.50 -> 192.168.1.101 ) , or will the asterisk just setup the call and will the audio stream go directly from phone to phone? ( audio stream goes 192.168.1.100 -> 192.168.1.101 )

a phone makes a call accross the sip trunk will the audio go through the asterisk ( audio stream goes 192.168.1.100 -> 192.168.1.50 -> 194.144.23.45) will the asterisk just setup the call and will the audio stream go from 192.168.1.100 -> 194.144.23.45 )

I would think it would do a connect between the phones and not hold the line, so to speak. I’m no expert by any stretch, but from what I know of other apps that is how it works.

Imbus

Technology evangelist for automated phone technology - IVR - Predictive Dialers - Appointment Reminders - PBX - Auto Dialers