On Tue, 31 Mar 2009 16:12:45 -0400, Mike wrote: >Maybe something like that could be done by using "set groups" and counting >the number of calls, and at a specified threshold (i.e. 6 simultaneous >calls) you`d specify g729 for new calls. > >Shifting an ongoing call might be impossible though. > >Mike
I'd not expect to shift an ongoing call. But if you know that your IP connection is constrained then when you have x calls in progress you might benefit from forcing any additional new calls into G.729. At least until your simultaneous call volume drops a bit. I image that most people who use G.729 simply use it all the time. But then, why take the quality hit if your call momentary volume is low? Michael > >> -----Original Message----- >> From: [email protected] [mailto:asterisk-users- >> [email protected]] On Behalf Of Michael Graves >> Sent: Tuesday, March 31, 2009 11:08 >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: [asterisk-users] dynamic codec preferences >> >> Has anyone here ever had the occasion to setup a system that would >> dynamically alter it's codec preferences based on trafffic? That is, >> presuming that the system is on a limited bandwidth connection is would >> start to prefer a compressed codec as the call volume increased? >> Perhaps shifting from G.711 to G.729? >> >> Michael >> -- >> Michael Graves >> mgraves<at>mstvp.com >> http://blog.mgraves.org >> o713-861-4005 >> c713-201-1262 >> sip:[email protected] >> skype mjgraves >> fwd 54245 >> >> >> >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > >_______________________________________________ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Michael Graves mgraves<at>mstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[email protected] skype mjgraves fwd 54245 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
