On Thu, 16 Dec 2004 14:51:53 -0600, Matthew Boehm <[EMAIL PROTECTED]> wrote: > Here is the setup: > > Phone A (in NYC) on own bandwidth. > Phone B (in LA) on own bandwidth. > Asterisk box in Houston,TX on own bandwidth. > > Both phones contact asterisk to register. Not much bandwidth used for this > as it is a few packets every hour or so. > > Phone A calls Phone B. Phone A sends a call request to asterisk and asterisk > calls phone B. Both phones are connected and both people are talking. > > Is all of the data/voice comming from phone A going into asterisk box and > then from asterisk box to phone B? If so, then using g711, phone A would > send/recieve 64Kbps to/from asterisk and phone B would also send/recieve > 64Kbps to/from asterisk. Asterisk would then be sending/recieving 128Kbps > for this one call right? So with 1 T1 you could only get 12 calls going > right? > > If I use canreinvite=yes on both phones, will phone A connect to phone B > directly therefore lowering the bandwidth usage in/out of the asterisk box > right? > > If so, what is the "signalling" bandwidth usage in/out of asterisk in this > case? Even if the phones are connected directly to eachother, they still > have to pass some data to asterisk so asterisk still knows that the call is > up and has to know when the call goes away. We need to know this bandwidth > usage on a T1 because lets say it was 10Kbps, you could actually do a bunch > of calls on 1 T1 provided that all phones use canreinvite right? > > Thanks, > Matthew > > _______________________________________________ > 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 > The answer is yes. If the a reinvite is issued then * is out of it but stays in there for the signaling. look at the following: http://www.voip-info.org/wiki-Asterisk+SIP+media+path http://www.voip-info.org/wiki-Asterisk+sip+canreinvite http://www.voip-info.org/tiki-index.php?page=Asterisk:%20Letting%20SIP%20clients%20connect%20directly _______________________________________________ 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
