On Dec 10, 2007 1:17 PM, Jerry Geis <[EMAIL PROTECTED]> wrote: > Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a > 64 bit 4200+ box > would there be any noticable lag or delay to bring each one of them into > a PAGE mode. so one speaker can talk out on all 230 SIP clients for a > message. > > Would this work? > > Thanks, > > Jerry >
I would also be really concerned about the ability for the NIC to serve up all of those RTP streams... 50pps x 230 = 11,500pps It would be nice to have some support for RTP multicast or something. Obviously this would require changes in Asterisk AND support in each phone, but it would be really cool. I think I've seen some Linksys/Sipura devices support it. -- Kristian Kielhofner _______________________________________________ --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
