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[asterisk at net153.net]

I am thinking about using my asterisk server to host a conference with
about 12 other people from around the USA. Bandwidth issues aside, will
this work or will all the different latencies cause issues? Yea I know,
I could just "try it and find out" but it is going to take alot of time
to get everyones schedule to line up, I don't want to go through the
trouble if I will just be disappointed.

Thanks,

Sam

[bwentdg at pipeline.com]

The 2 big questions are:
-Are all participants using QoS end to end ?

-Are all of them using the SAME CODEC. As the amount of Transcoding goes up,
the work on the * box goes up and can be a problem.

Sam wrote:

[asterisk at net153.net]

Actually I think they will all be calling in using regular pstn phones
and cell phones.

Sam

Al Baker wrote:

[covici at ccs.covici.com]

12 people is nothing -- I do 20 regularly -- however you may want to
have them come in as muted or tell them to mute themselves, because
the latency can cause very severe echoes if they are on a speaker
phone or cell phone.

on Sunday 06/08/2008 Sam(asterisk at net153.net) wrote

[Adrian.Marsh at ubiqui...]

I've got to agree.. I've never given it much thought either...

All of my calls are SIP/IAX based, coming in from the PSTN from a peer
like that too..

I've never tracked the total number of conference users... But I'll bet
we've hit at least 10.. And I've never seen the CPU go above 10%.. And
that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it
will be setup-specific.. So I would look at your CPU and memory stats,
and run some tests and monitor that..

A.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of John
covici
Sent: 08 June 2008 16:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MeetMe Limits

12 people is nothing -- I do 20 regularly -- however you may want to
have them come in as muted or tell them to mute themselves, because the
latency can cause very severe echoes if they are on a speaker phone or
cell phone.

on Sunday 06/08/2008 Sam(asterisk at net153.net) wrote > Actually I think
they will all be calling in using regular pstn phones > and cell
phones.

[astmattf at gmail.com]

Hello,

We routinely run meetme with over 140 ULAW channels connected to 70
meetme rooms with no issues on an Intel Core 2 Quad core CPU.

The major factor in capacity would be your CPU and RAM capacity. If
you have at least a base-level P4 you don't need to worry about 12
participants.

MATT---

On 6/8/08, Adrian Marsh <Adrian.Marsh at ubiquisys.com> wrote:

[stotaro at totarotechn...]

Matt,

Could you share the CPU usage, memory, and load average in the
scenario you describe? What type of ULAW channels
(SIP,DAHDI,IAX....), or does it not matter?

Thanks,
Steve Totaro

On Sun, Jun 8, 2008 at 5:29 PM, Matt Florell <astmattf at gmail.com> wrote:

[astmattf at gmail.com]

Hello,

The load is usually quite high because this is VICIDIAL inbound call
center traffic with full Asterisk-based recording. On a system with
70-80 Meetme rooms running with 2 participants each doing full
Asterisk-based recording in each Meetme room the loadavg stays between
2.00-4.00 on a Quad-core Intel core 2 Quad processor with 4GB RAM. I
have three systems like this in place at different call centers and
the load is consistent for all three of them. Usually we put less load
on a single server, but these were inbound-only scenarios which is
less load than outbound.

MATT---

On 6/8/08, Steve Totaro <stotaro at totarotechnologies.com> wrote:

[astmattf at gmail.com]

Forgot to address your second question. DAHDI, that's a good one

The channel type doesn't seem to matter. One has all agents on Zap
channels through channelbanks with all calls coming in over IAX and
monitoring done through SIP. One has all SIP agents with all calls
coming in over SIP trunks, and another has SIP agents with calls
coming in over Zap T1 channels.

MATT---

On 6/8/08, Matt Florell <astmattf at gmail.com> wrote:

[gordon+asterisk at dro...]

On Sun, 8 Jun 2008, Matt Florell wrote: