Hi,
I Have a problem here, if anyone know a method to avoid please tell me .
Using * with the option canreinvite=yes i can in theory tell to my * box, send RTP Packet directly from one
Sip device to another one, then "In Theory", i will not use my own internet connection.
So this mean that will a lower connection something like "512/512kbps", i can have lot's of channels connected
and using my * just to bridge then .
Thats correct ??
But if all people is under a NAT ??
Like 2 sip devices using * box connect over my * box but that two is under NAT ??
Will work anyway ? All RTP Packets is flow using their connection ? Not mine ?
If anyone understand this nonsense question please help me ;)
Thanks alot !!
Carlos.
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users