[Asterisk-Users] How does Asterisk handle connecting two IP end points?
When a connection is carried between two IP end points, does the Asterisk server incur CPU usage to pass the voice bearing circuit between the two end points? Is it possible to have Asterisk setup the call but hand off the voice traffic to be handled directly between the two end points? Thanks! Chip __ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does Asterisk handle connecting two IP end points?
IIRC asterisk by default will not participate in the call between two SIP phones.. It will help establish the session to the correct UA and then have nothing more to do with it unless the call is transferred to another UA in which case Asrerisk will again be involved in setting up the call.. So no when 2 SIP UA'a are connected there should be no CPU load on the Asterisk server.. Hope that helps.. When a connection is carried between two IP end points, does the Asterisk server incur CPU usage to pass the voice bearing circuit between the two end points? Is it possible to have Asterisk setup the call but hand off the voice traffic to be handled directly between the two end points? Thanks! Chip __ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does Asterisk handle connecting two IP end points?
On Thu, Jul 03, 2003 at 03:57:09PM +, WipeOut . wrote: IIRC asterisk by default will not participate in the call between two SIP phones.. It will help establish the session to the correct UA and then have nothing more to do with it unless the call is transferred to another UA in which case Asrerisk will again be involved in setting up the call.. Asterisk will be handling the signalling for the call, not the voice stream. If you look at 'show channels' or 'sip show channels' while the call is up you will see that Asterisk is aware of it. Running tcpdump will show you that the phones are still talking to Asterisk. So no when 2 SIP UA'a are connected there should be no CPU load on the Asterisk server.. Well there is a minimum amount just for keeping track of the call, but per call is very low. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
