Maximum Number Of Users

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Hello;

Can someone advise me what is the maximum number of users (IP Phones) that can be supported by asterisk 1.8 or later?

Regards Bilal

14 thoughts on - Maximum Number Of Users

  • You can have tens of thousands of phones as long as no one makes or receives any calls J. The better question to ask is how many concurrent calls have people been able to make. The quick answer is it depends on many things.

    John

    From: asterisk-users-bounces@lists.digium.com
    [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of bilal ghayyad Sent: Wednesday, December 18, 2013 9:46 AM
    To: Asterisk Users Mailing List – Non- Commercial Discussion Subject: [asterisk-users] Maximum number of users

    Hello;

    Can someone advise me what is the maximum number of users (IP Phones) that can be supported by asterisk 1.8 or later?

    Regards

    Bilal

  • The better question is, maximum number of users (IP Phones) can your hardware support. I have * deployments with 300-600 phones. – works fine. though concurrent calls has never seen more then 244. Also at this point I
    have to ask, for this to be any concern to you, you must either A, Make tons of internal calls. OR B have multiple T1’s/lots of sip channels?

    Regards,

    Keith Sloan Voice Operations Center Vianet
    705-222-9996 X7203
    1-800-788-0363 X7203
    keiths@vianet.ca

  • How did the system behave with 244 calls? I’ve been able to make 1,024
    concurrent faxes (which tend to use more resources than audio calls) in the lab. The problem I had was after the faxes were transmitted, things couldn’t keep up and kept dumping core. Two things were going on, (1) the CDR was written to MySQL and (2) a FastAGI script (I use the AGISpeedy PERL package)
    to write a log entry also to MySQL. I tried switching the CDR’s to sqlite and that seemed able to keep up, except that its concurrency issues were a problem. If MySQL is the problem, I could probably optimize it better, but it doesn’t explain the Asterisk core dumps. It might be related to the number of FastAGI scripts running, I’m not sure at this point.

    Regards;

    John

    From: asterisk-users-bounces@lists.digium.com
    [mailto:asterisk-users-bounces@lists.digium.com] Hello;

    Can someone advise me what is the maximum number of users (IP Phones) that can be supported by asterisk 1.8 or later?

    Regards

    Bilal

  • To be honest I am not sure, I pulled the data from a cacti graph shortly before posting my reply. I imagine it all ended well, as I don’t recall hearing complaints on quality. I have noticed the core dump issue mixed with * and SQL a few months ago. We had about 120-150 users in a conference, that seemed to be okay, heard a complaint about a periodic sputter in audio, but nothing to serious. Issues came once the call was done, ~150 users all hanging up at once. SQL was VERY upset with that, causing * to choke. Though I think it was how we handled the deployment to get it up quickly, and may have been able to prevent this, if tested better.

    Keith

  • We have a machine with a quad core ‘Intel(R) Xeon(R) CPU E5-1410 0 @
    2.80GHz’ running asterisk 11.2-cert with ingress and egress all sip. Fastagi running as a daemon (written in perl) performing cdr updates at call start, answer and call end together with a query when a call comes in to get information on what to do with it. Local Mysql 5.5 database. With all this we easily handled 450 simultaneous calls with many of them also performing call recording. I think we typically had 60% idle on each processor so the box could be pushed a lot more.

  • Have you ever checked out the app_konference module? You can check it out here. http://sourceforge.net/projects/appkonference. I have a customer who routinely hosts 100+ users in a conference without issue. We’ve had very good results so far. We’re hoping to eventually hit 500+ users in the future with a simple hardware upgrade and a better SIP provider.

    Regards;

    John

    From: asterisk-users-bounces@lists.digium.com
    [mailto:asterisk-users-bounces@lists.digium.com] Hello;

    Can someone advise me what is the maximum number of users (IP Phones) that can be supported by asterisk 1.8 or later?

    Regards

    Bilal

  • Yeah. I started looking at this a few weeks ago. I am going to do a trial deployment in the new year. Where are you located in the world?

    Regards,

    Keith Sloan Voice Operations Center Vianet
    705-222-9996 X7203
    1-800-788-0363 X7203
    keiths@vianet.ca

  • Central Maryland, USA. About an hour NW from Washington, DC.

    John

    From: asterisk-users-bounces@lists.digium.com
    [mailto:asterisk-users-bounces@lists.digium.com] Hello;

    Can someone advise me what is the maximum number of users (IP Phones) that can be supported by asterisk 1.8 or later?

    Regards

    Bilal

  • Hi Bilal,

    Assuming you have the latest hardware, sufficient memory, cpu, etc… The key to determine the maximum number of users comes down to the office type, RTP path, network interface, and primary codec used.

    First we need to determine the over-subscription rate, how many people will be using the phones at any given time.

    For a call center, the ratio is 1:1. For a normal office, the industry standard is 4:1.
    {This ratio is also used to determine the number of PSTN channels you will need too}

    Will the PSTN connections be Digium card(s) in your server or external gateway(s)?
    Assuming Diguim card(s), the RTP will be going through your server.

    Determine the network interface. 10/100/1000baseT
    Then we need to consider the largest codec used, and divide the available bandwidth by the typical packet size.

  • Thats fine for calculating how many users a particular speed network connection can cope with. 640 concurrent calls on a 100Mbps connection is doable on a decent machine as long as you are not doing much codec translation.

  • The number of devices and concurrent calls is dependent on many factors. Dialplan complexity, new call rate, features enabled, and transcoding all play a factor in these numbers.

    To give you an example I have a Dell R710 with two quad core E5520
    processors running Asterisk 1.8 and FreePBX 2.11. I have around 1,000 SIP
    device registrations, 50-80 concurrent calls for the majority of the day, and a total of 8-10k calls processed per day. A few times a week I will see the last minute load at 20 and the 5 min load at 7. This seem to happen when there are a high volume of new calls as the FreePBX dialplan is complex.

    Ryan

  • Thanks a lot for the help from the all.

    Without using PBX functionalities (like conference or pickup and so on), only basic calls. So how many concurrent calls can support?

    The idea is, we need to use Asterisk with ISP which will be service provider for sip calls for the subscribers, and the asterisk should be connected with E1s to do calls within the country.

    The registered users will reach up to 100 000 users and the concurrent calls will reach up to 2000 or 3000 calls.

    Appreciate the kindly advise. Regards Bilal

    You can have tens of thousands of phones as long as no one makes or receives any calls J. The better question to ask is how many concurrent calls have people been able to make. The quick answer is it depends on many things. John

  • You need to decide which codecs you are going to allow to be used on the SIP side. As you are connecting to E1 then the standard codec would be g711 alaw or ulaw. You could force the SIP side to use the same codec but it uses about 100Kbps of bandwidth so quite a bit higher than other codecs and some customers may prefer to use a higher compression codec so they dont have to have a faster internet connection. If you start allowing other codecs then asterisk will need to perform codec translation which consumes CPU.

    You will be limited by the E1 cards you can fit into asterisk. Generally its best to keep to a maximum of 2 PCI cards in a server as they each generate a lot of interrupts. A couple of Sangoma A116 will give you 32
    E1 circuits which can handle 960 calls. That may well be pushing the server to its limits of what it can do even if it is just handling basic calls. So you will want 3 of these systems and preferably four as you will want some redundancy.

    You would want something like opensips at the front end on a couple of redundant servers. You can use the load balancing module in opensips to spread the calls evenly between the available asterisk servers.

    You should also have a think about whether you wish to use asterisk at all. Opensips will do the load balancing that you need and can perform the billing actions aswell. If all asterisk is doing is performing sip to E1 conversion there may be hardware solutions which will be more cost effective and require less maintenance.

  • The IRQ distribution should be checked on a multi-processor system. I typically have 3 Sangoma cards in low end Xeon servers and there aren’t any problems.

    Recently, I needed to patch irqbalance to evaluate the numa node values because some somewhat older versions of CentOS (6.3, 6.4 initially) do not set certain links inside sysfs. If there is some interest, I could publish my small patch.

    jg