17 Mar
2008
17 Mar
'08
3:03 a.m.
It's my understanding that when I call from one SIP address to another that Freeswitch manages the invites and then directly connects the two sip devices, such that freeswitch is no longer involved in the conversation. Is this how it is supposed to work? The reason for my question is that the rtp traffic is going through my freeswitch ip address at all times. It seems that both ends are speaking g711u, so I don't believe that any transcoding is going on. I must be missing something because having all data flow through Freeswitch would not be scalable and I know that one of the big differences that Freeswitch offers is scalability. What am I missing....and I know that I am:) Kurt