On Dec 12, 2007 9:41 AM, Russell Bryant <[EMAIL PROTECTED]> wrote: > BJ Weschke wrote: > > Jerry Geis 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. > >> > > I would have some serious reservations throwing this many clients into > > an app_meetme room which is the foundation layer for the page > > functionality. > > > > Well, it may be ok, especially given that the 230 clients are all marked as > listen only. There isn't any mixing going on at all. > > However, there is almost certainly going to be some lag that you may not be > happy with. What happens is that you are spawning 230 threads to make > outbound > calls and connect them to MeetMe, all at the same time. This process is far > from instantaneous. :) > > I would also be concerned about the effects that this spike in extra > processing > would have on the quality of any existing calls on the system. > > But, as with most things, the only way to know for sure is to do some testing. > > -- > Russell Bryant > Senior Software Engineer > Open Source Team Lead > Digium, Inc. >
Russell, What are your thoughts on SIP/RTP multicast, if any? It's been discussed before. Seems like a great solution for paging (f the phones support it). Anyone interested in a bounty? -- 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
