On Mar 16, 2008, at 10:03 PM, Kurt Marasco wrote:
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?
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall
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.
Well in some cases you can do p2p but since no transcoding is going on the only limit is the amount of context switches your machine can take on. I know you can do way more than you think doing media thru FreeSWITCH.
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.
Actually it can scale with media just not as far. If you set the bypass_media=true variable before you bridge we'll get out of the media path. There are instances where you want to always be in the media path. This paired with late negotiation can give you access to the SDP and even allow you to choose the mode on the fly. /b
What am I missing....and I know that I am:) Kurt
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