Asterisk Conference/Meet Me for 50 Users

torontob

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#1
Hello,

Do you know of any white papers on Asterisk Meetme/conference features or similar open source projects that has in-depth details of capacity and feasibility for 50 user conference? Client likes to have Web GUI feature access for kicking, banning, inviting, muting, and reports. Other than that, the users will call in and I can have the system to record their phone number (DTMF), and then when conference time comes up the system will call with 3 tries to join them to the room. Hence it is all outbound calls and everything is done by SIP trucking. Bandwidth is not a problem and the server will be co-located.

I am wondering how feasible the project is with Elastix and if there are those who have had loads of 50 users with an Elastix server?

Thanks
 

Chilling_Silence

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#2
Yeah 50 calls is nothing, you can do that quite easily. I'd highly recommend iLBC / g729 and possible IAX to your provider if you're going to pump that many calls.

Ive got a Dual-core 1.6Ghz Atom that I figure can handle 90+ calls when not transcoding :)
 

torontob

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#3
Great input. Thanks.
1- Have you had a lot of users on conference call in one single room? I need 50 users in 1 room.
2- Why do you recommand IAX vs SIP and why g729? Just bandwidth concern?
3- If I understand it right, iLBC is free but is it as good as g729? Quality is important for us.
4- What do you use for controls? Web-MeetMe? IS there a quick install to Elastix on WebMeetMe?

Thanks again
 

Chilling_Silence

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#4
1) Nah havent had that many in a single room, but I dont know of any limitations to the number of users you can have in a single room.
2) IAX is nicer because of the ports, it uses a single one, means less open in your routers state table, which in-turn means potential greater reliability :)
3) I'd go iLBC over g729 just coz its nicer with packet loss, and the quality is equal to (or better IMO but maybe its just my funny ears) g729 when no packet loss, but when it starts losing packets its quality is much higher than g711 or g729 :)
4) No controls, just the setup in FreePBX, tell users to press * and mute themselves :)
 

torontob

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#5
Thanks. I like the port explaination. Would help if ther server was not a dedicated one or the only server in a complex.

Has anyone got WebMeetMe to work with Elastix? I see some really lengthy documents on how to get it running which I think is a waste of time provided Trixbox has it made already and it should much more reliable than what I would achive in the next six month. I'd appreciate to see docs from people who might have implemented it with Elastix.

Thanks again Chilling_Silence
 

Chilling_Silence

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#6
It really depends on your QoS, but definitely dont run that in a VPS, use dedicated hardware, something semi-beefy should suffice. Dual-core 2Ghz Pentium upwards ought to be way more than sufficient as long as you're not transcoding.

For bandwidth, see here:
http://site.asteriskguide.com/bandcalc/bandcalc.php
50 calls for iLBC 13.3kbps requires 956kbps (Approx) bandwidth when using a trunked IAX service (Ive specified ADSL PPPoA as the transport), whereas a SIP service would use 2204kbps.
50 calls for g729 (8kbps) requires 686kbps when using trunked IAX, or 2226kbps for SIP
50 calls for g711 (64kbps) requires 4274 for trunked IAX, or 5565 for SIP.

You'll preferrably have some sort of QoS so that if theres something else on that connection (Another Email / Web server perhaps) that the Voice isnt interrupted.

Looking at installation instructions for WebMeetMe here:
http://areski.net/Web-MeetMe/about.php?s=0
It looks relatively easy... ?
Installation Instruction

* 1) cp phpagi.example.conf /etc/asterisk/phpagi.conf
* 2) vi /etc/asterisk/phpagi.conf
Change the file according to your manager settings
* 3) Enjoy the Web-MeetME UI !
I told you it was easy...
* 4) Don't forget to make a donation ;)
you will feel better after this...
 

dicko

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#7
chill:
Love the bandwidth calculator link,
Thanks a bunch (+1)!
 

Chilling_Silence

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#8
Cool mate, no worries B)

DEFINITELY comes in handy for planning bandwidth, on shared AND dedicated connections :)

Lucky for me, our local ITSP not only supports iLBC, g729, but is happy for us to chop & change between SIP & IAX2 as we please :)
 

torontob

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#9
What is IAX2 Trunked vs IAX2 service? Is the first one a direct connect to an Asterisk Server vs the other a transcode between IAX2 and SIP on the provider side?

Thanks the for the URL.
 

torontob

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#10
Also, can you guys recommand an ITSP which would do dedicated hosting and has good A-Z rates for the purpose of this conference room? Don't mind it to be in USA. I am in Canada. Not much choice here with ITSPs and different codecs.

Thanks
 

Chilling_Silence

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#11
Have a look here under the "trunking" header: http://www.voip-info.org/wiki-IAX
Basically they share the same packet stream and cut down on bandwidth
Its the "trunk=yes" that affects it

http://www.voip-info.org/wiki/view/IAX+versus+SIP

Wish I could help with the ITSP but I'm from New Zealand, not much help there ;)
If you're going 50-odd concurrent calls, its probably worth applying for a wholesale account with an ITSP.
 

ramoncio

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