300 users using the same extension to register, simultaneously? (mobile SIP client)

Hello everyone,
I’m working on a slightly unusual configuration, and any advice would be GREATLY appreciated:

we’re doing speech broadcasting and I am trying to make the following system work:
multiple attendees (around 300) register with a SIP client (Zoiper, XLite, etc.) on their smartphones. Then they dial out into the “conference” and listen to the broadcast.

Now, for various reasons I would like to avoid the hassle of creating 300 different extensions and providing different SIP registration info to each user separately.

So the idea is to create only ONE extension, let say 1111. Every user registers with exactly the same extension 1111 on their smartphone (that’s 300 users on one extension). Users dial out into the conference and listen to it.
When the broadcast is finished I will change that single extension password to unregister all the attendees (every broadcast will have new attendees every time, so I need to make sure old attendees are unregistered).

I have now ran a test with 3 smartphones logged in as the same extension, and dialled into a conference. Works flawless so far.

Do you think this will work for 300 users as good as it does with 3?

I do not need any “incoming call” functionality, users only dial out to the conference extension.
I’m running the latest FreePBX Stable-6.12.65 with Asterisk 11, in “extension” mode.

Perhaps, if you are using g711 you will need an unimpeachable symmetric 24Mbs connection between your server and where your extensions are at (scale up and down between bandwidth/codec/cpu power to conference all those connections) at 300 calls you are seriously pushing your ulimits on a standard install, compounded if you transcode.

All the attendees are in the same building as the server, so the total bandwidth requirement should not be a problem.

Regarding CPU, etc:
I plan to run it on i7-4770t with 16GB RAM and 250 GB SSD.

Then “suck it and see”, my guess is you are close to “out of dilithium crystals” here though.

Dicko, are you saying that basically running multiple users on one extension is not a problem, right?
If so, why do you think that 300 concurrent calls should be a problem? General Asterisk stability? Or the server is not powerful enough? (I could run two boxes instead of one then)

Very far from that, I am just (re)saying:-

“Then “suck it and see”, my guess is you are close to “out of dilithium crystals” here though.”

If you want that sort of scale , perhaps look at Kamailio or even FreeSwitch as your media server.

You must construct additional pylons.

Thank you guys for the inputs.
The reason I try to stay away from Kamailio and FreeSwitch is the smaller community, lack of well developed GUI, lack of integrated support for ddns, and more complicated installation (unfortunately I am not an IT developer and quite new to VoIP too).

But if you say that 300 concurrent calls is a problem, then I wonder why does Voip-Info.org reports something like 1500 problem-free concurrent calls (no transcoding)? Any personal experience here?

And again, if I run not 300 users but 100, to reduce the load – is running multiple users on one extension (as in my original question) going to be a problem?

The problem is not the concurrent calls so much as the ability to mix and balance 300 legs of a conference bridge.

But really the only easy to know is, again, to just “test it”. but I suggest that you should really be looking for a multicast solution.

p.s. What do you mean exactly by

“running multiple users on one extension”


Yes unless you switch to user/device mode. As soon as you register an extension from one IP, it will force the other registration to expire. Otherwise, I would recommend the bulk extensions module; pick a range of extensions to use for the conference and throw them into the PBX…

The company I work for does teleconferencing like this, and it’s a matter of system resources. The server handling the calls has to do a LOT of work and 300 people is a potential capacity issue. We have a beefy system and we can only process something like 80 calls a second with a max capacity of 2,500 calls per site at any given time.

I’m assuming the reason you don’t want to set it up with a DID to the conference app is because of the cost of the call at 300durationprice/min? That would be $900 for an hour call at 5¢/minute (18,000 minutes). Our service ranges from 5-12¢/minute and depending on how much usage you plan on…so I totally get looking to save money there.