400+ User Conf Bridge

zigg.z

New Member
Joined
Feb 24, 2010
Messages
58
Reaction score
0
I am trying to see if I can come up with a server for a non profit organization, and I have a few questions.
I would like to make a server (or set of servers) that would handle 400 callers calling in for a managed and "quiet-seating" conference.
I would like some input from anyone who can shed light or ideas of any kind. I have set up several systems that have been quite dependable, and are going pretty fair, and now I was asked if I could help to make a 400 user conference system for the non profit with a few special parameters.

I encourage and ask for input from anyone who understands this.

I would like to know some of the following:

•The conference would need to be administrated by a whitelist of CID numbers allowed into the conference, and if their CID is not matching, they are not allowed into the conference.

•What is the best payload OS (and I would assume 64Bit?)

•What is the best server hardware? (or set of servers)
Dell? HP? Xeon Quad core? how much ram?

•What is the best timing device for a conference system like this?

•What codec would be used?

•What is the rule to calculate the number of calls for the bandwidth available to handle this traffic over SIP? I just ran a speedtest, with 7.8mb download, and 10.8mb upload-- I am sure I will have to set up another server in another location to take half the load, and I can put one in a 50mb connection, but now I will need to know how to interconnect the two, and if it goes bigger than that, how to interconnect the third for a managed conference, unless they must be 3 separately managed conferences, I would be good with that.

•All callers must automatically be muted when calling in, unless an administrator code is entered

•Can someone come up with a way to import a CSV file of CID numbers to administrate the conference? If it is something that someone would like to do for hire, I would like to know the price of anyone that could help me put this together.
 
Seems like most suggestions in this area lean toward using a third party service unless your going to be doing this all the time.

Some notes I saw suggested using :

http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference

instead of the meetme functionality. Users reported setting up conf with 150+.

This may also be a help in calc bandwidth required:

http://blog.asteriskguide.com/bandcalc/bandcalc.php

This provides background and other suggestions:

http://www.freeswitch.org/node/100


I'm sure there will be other comments.....

Brian
 
Hardware: Quad-core Xeon, 2G RAM should be fine.

Network: 50M symmetric internet connection

Timing Source: Non-populated FXS/FXO card should work great, assuming you're doing this on PiaF

Note: something like this should probably be done via a custom asterisk install... not really PiaF. IMO, specific requests like this should be tailored to ensure reliably.
 
Bounty?

If someone would be interested in building this, I would entertain quotes.
I will need for sure all listeners to be default silent, and the whitelist via csv file is super important
 
ZIG,

It could be built, but before I ran numbers I would need a better idea on the number of calls per month and the length of the calls. You are talking sizable bandwidth and with out better numbers it would be hard to size / price.

I think you migt be better off with a a package from someone like this;

http://www.freeconferencecallhd.com/

----------------------
 
approx figures

I am anticipating 350 callers average, so planning for 400+ would be ideal.
If it needs to be spread across two or more servers for bandwidth needs etc, that might be part of the plan.

I am anticipating three calls per week, for upwards of 2 hours each, upwards of 350 users, calling in on cell phones and landlines
 
From my math, 350 concurrent calls at 100 kbps each works out to 35 Mbps bandwidth. Unless you use a better codec which means more work on the CPU end
 
Great, I am not opposed to setting up two servers.
I don't know what someone might think this is worth, but I will keep the fire lit until we get something resolved.
They have used third party conferences before, and want it all in house.
What will it take to get this conference system compiled with the above parameters?
 
Okay,

Please define in house ?? 350+ calls TDM is 15+ Pri links at about $500 a month not to mention the capital expense and Sangoma or Redfone termination hardware.

This really should be placed in a CoLocation Facility which may NOT EQUAL in house. An I am guessing that you are in the 2.5K to 5.K a month for the CoLo plus the bandwith (IP) or PRI Charges, plus usage from the carrier, plus the capital charges to build it all.

If it's a political decission then this might fly, if it's a business case based logical decission I don't know how you could justify it VS a service provider.

------
 
Colo isn't that much money... I have a 120 sq/ft cage w/4 racks for around $2200/mo after power costs.
 
Interested to hear what you eventually wound up doing zigg -- please give us an update if you have the time & inclination. :)
 
(look ma) Mass conference , no new software

I would like to make a server (or set of servers) that would handle 400 callers calling in for a managed and "quiet-seating" conference.


By "quiet-seating" , I imagine you mean that those coming
into the conference only hear the ongoing speakers, and
do not speak themselves.

As a programmer I've not yet played with the concept of a
mass broadcast conference/meetme ,but the mention of the
idea naturally starts the mental wheels turning.

It is likely one can accomplish this with no new software at all.

Just sorta spit balling here, but looking at the goal
from my vantage point as a programmer, it sorta
looks like fancy live "hold music".

As anyone dialing a customer service center can attest
Asterisk seems to do really well at sharing hold music
among a large number of listeners and affording different
"hold music" to different contexts.

There is a special file type in Linux that while
looking like a normal file (such as mohLiveLecture.wav) ,
can be configured to act as a shared FIFO buffer.
In a way this works sort of like the delay used in live
call in radio shows.

In this fashion it should be possible to simply
use normal asterisk features to "monitor" a conference
to the magic mohLiveLecture.wav file, while the
conference listeners are hearing mohLiveLecture.wav "hold music" with but a minor delay.

Since this is "quiet-seating" the delay would not be a problem.

If one wanted to allow an audience member to speak,
they would simply be "transferred" from the "hold music" to
the actual "conference", which to them would seem like going
on-air on talk radio as they transition from hearing the second or two
delayed broadcast to hearing the conference room live.

If one ever decided to do that audience call in thing ,
adding a one or two second bumper ("Your up next recording") would distract the caller
long enough that they would not even perceive the slight delay
when going live. Those actually in the conference room should experience
no delay at all with their experience being a regular conference call.
Only the audience simply listening to the conference call would have the slight
delay, but because they are not speaking , they never even know there
is a delay at all.


Configured properly , remote servers could probably "mount"
this same "hold music" mohLiveLecture.wav file thus
distributing the conference across as many servers as needed


The really cool things here are

  • Likely no new software is required, just configuration.
  • The features being leveraged are both mainstream and of long standing

Now as I said , I'm just spit balling, but all the
pieces are built into the OS and Asterisk , so
it SHOULD just be a matter of setting it up.


So basically the users perceive the situation as one giant conference call,
but in reality the conference room is only conferencing those speaking while
the attendees simply listening to the "hold music" which happens
to be a slightly delayed recording of the conference.


I could probably improve on this once I refreshed my memory on some
optional details, but if I recall correctly I might even be able to skip the "monitor"
and maybe adjust the character of an existing file to get the FIFO going (even better performance).







--Doc
 
I think I remember something on voip-info.org about using shell pipes + mplayer to create a streaming music on hold context that people could dial into. Yup, look here and scroll down to the section that says "Example using asx (mms://)(.wmv) streams"

I have not tried this myself but -- worth a shot. I think mplayer can also be made to play audio coming into a JACK interface (e.g. physical audio line-in connection like a microphone or line out off a mixer).

musiconhold.conf
Code:
[default]
 mode=custom
 dir=/var/lib/asterisk/mohmp3-empty
 application=/etc/asterisk/mohstream.sh


/etc/asterisk/mohstream.sh
Code:
#!/bin/bash
if -n "`ls /tmp/mayakpipe`" ; then
     rm /tmp/mayakpipe
fi 
PIPE="/tmp/mayakpipe" 
mkfifo $PIPE
cat $PIPE &
sleep 3
mplayer -cache 8192 -cache-min 4 mms://stream.rfn.ru/mayak -really-quiet -quiet -ao pcm:fast -af resample=8000,channels=1,format=mulaw -ao pcm:file=$PIPE
rm $PIPE
 
Well, I did not look anything up, but rather just
gave my reaction from what I happened to recall
about whats available.

The half of the music on hold solution you cite
would (have not read it yet) ( Just read it) probably deal with the
MOH side of it.


The other side is probably solved with one of two approaches.
One would be to use the "monitor" features of asterisk to get the conference
dumped into the MOH file. If I'm recalling properly their might actually be a file
already being created for the conference by Asterisk during the mixing, but I'm not positive.

If that file does exist, and is or were converted to a pipe, then
there would be no additional encoding overhead.

Once one does have the MOH pipe by whatever
means , having slave machines mount the file
shifts nearly all of the load in actually delivering the file to
those slaves.

In this fashion , one can (even dynamically) shift bandwidth
consumption and CPU load anywhere on a network or to
machines on the Internet at large as load demands dictate.

This means that the server where the conference
originates only needs enough resources to support
those actually speaking in the conference
plus the one FIFO buffer output stream to a mount somewhere else.

This also means that some attendees can elect
to listen to the same stream through typical internet audio players
by simply mounting the stream in an appropriate location where
internet users can click on it.

Add something like "click to call" and you have a live call-in show
that can use both telephones and the internet interactively with
even thousands of listeners.

The flexibility of named pipes makes almost anything
possible. This flexibility also means there are more options
in keeping costs down.
 
This really should be placed in a CoLocation Facility which may NOT EQUAL in house. An I am guessing that you are in the 2.5K to 5.K a month for the CoLo plus the bandwith (IP) or PRI Charges, plus usage from the carrier, plus the capital charges to build it all.

Old thread, but not that old...a co-lo for this type of setup would be nowhere near 2.5K to 5K a month. You're looking at a couple hundred bucks tops.

Plug: www.cingularhosting.net =P
 
You're looking at a couple hundred bucks tops.

I agree with your estimation of the costs. If one
manages to spend even as much as $200 , I would
think the injury self inflicted for not shopping around.

The biggest expense will probably be the time
someone spends doing the configuration work, but
of course that would be a capital expense, not
a reoccurring expense.

I can understand the confusion though as there are
still a lot of old school businesses trying to sell expensive pipe,
even when they know there are cheaper alternatives available
to the customer.

I've even seen hosting services that will sell you bandwidth on demand,
by the day, so one could buy bandwidth just the day of the lecture
and but as needed, with the expense ending that day.

--Doc
 

Members online

No members online now.

Forum statistics

Threads
26,687
Messages
174,410
Members
20,257
Latest member
Dempan
Get 3CX - Absolutely Free!

Link up your team and customers Phone System Live Chat Video Conferencing

Hosted or Self-managed. Up to 10 users free forever. No credit card. Try risk free.

3CX
A 3CX Account with that email already exists. You will be redirected to the Customer Portal to sign in or reset your password if you've forgotten it.
Back
Top