[asterisk-users] asterisk 1.4 with around 230 SIP connections

[Kristian Kielhofner]

On Dec 12, 2007 9:41 AM, Russell Bryant <russell at digium.com> 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