VoIP Mailing List Archives

[bilmar_gh at yahoo.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

[keiths at vianet.ca]

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 (keiths@vianet.ca)




On Wed, Dec 18, 2013 at 9:45 AM, bilal ghayyad <bilmar_gh@yahoo.com (bilmar_gh@yahoo.com)> wrote:

[asterisk at voipbusine...]

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

[asterisk at voipbusine...]

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] On Behalf Of Keith Sloan
Sent: Wednesday, December 18, 2013 10:10 AM
To: bilal ghayyad; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Maximum number of users


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 (keiths@vianet.ca)




On Wed, Dec 18, 2013 at 9:45 AM, bilal ghayyad <bilmar_gh@yahoo.com (bilmar_gh@yahoo.com)> wrote:
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




--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

[keiths at vianet.ca]

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



On Wed, Dec 18, 2013 at 10:46 AM, Tech Support <asterisk@voipbusiness.us (asterisk@voipbusiness.us)> wrote:

[mailinglist+asterisk a...]

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.

On 18/12/13 15:46, Tech Support wrote:

[asterisk at voipbusine...]

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] On Behalf Of Keith Sloan
Sent: Wednesday, December 18, 2013 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Maximum number of users


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




On Wed, Dec 18, 2013 at 10:46 AM, Tech Support <asterisk@voipbusiness.us (asterisk@voipbusiness.us)> wrote:
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 (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Keith Sloan
Sent: Wednesday, December 18, 2013 10:10 AM
To: bilal ghayyad; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Maximum number of users


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

[url=tel:705-222-9996%20X7203]705-222-9996 X7203[/url]

[url=tel:1-800-788-0363%20X7203]1-800-788-0363 X7203[/url]

keiths@vianet.ca (keiths@vianet.ca)




On Wed, Dec 18, 2013 at 9:45 AM, bilal ghayyad <bilmar_gh@yahoo.com (bilmar_gh@yahoo.com)> wrote:
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




--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users








--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

[keiths at vianet.ca]

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 SloanVoice Operations Center
Vianet
705-222-9996 X7203
1-800-788-0363 X7203
keiths@vianet.ca (keiths@vianet.ca)




On Wed, Dec 18, 2013 at 12:05 PM, Tech Support <asterisk@voipbusiness.us (asterisk@voipbusiness.us)> wrote:

[asterisk at voipbusine...]

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] On Behalf Of Keith Sloan
Sent: Wednesday, December 18, 2013 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Maximum number of users


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 (keiths@vianet.ca)




On Wed, Dec 18, 2013 at 12:05 PM, Tech Support <asterisk@voipbusiness.us (asterisk@voipbusiness.us)> wrote:
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 (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Keith Sloan
Sent: Wednesday, December 18, 2013 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] Maximum number of users




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




On Wed, Dec 18, 2013 at 10:46 AM, Tech Support <asterisk@voipbusiness.us (asterisk@voipbusiness.us)> wrote:
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 (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Keith Sloan
Sent: Wednesday, December 18, 2013 10:10 AM
To: bilal ghayyad; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Maximum number of users


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

[url=tel:705-222-9996%20X7203]705-222-9996 X7203[/url]

[url=tel:1-800-788-0363%20X7203]1-800-788-0363 X7203[/url]

keiths@vianet.ca (keiths@vianet.ca)




On Wed, Dec 18, 2013 at 9:45 AM, bilal ghayyad <bilmar_gh@yahoo.com (bilmar_gh@yahoo.com)> wrote:
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




--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users








--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users









--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

[b.lavallee at globalta...]

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.

µ-law/A-law is roughly 80 kbps, so we can support 128/1280/13107 audio streams.
Divide that by 2 (just to be safe) and allow RTP in both directions. 64/640/6553

Now multiple the result by the over-subscription ratio. 4:1 = 256/2560/26212

So we see that the maximum number of users is 2560 for a normal office when there is a 100baseT NIC in your Asterisk server.
You would also need to have 640 channels (28 T1 PRI's) connecting to the PSTN.
Using SIP trunks to connect to the PSTN through the same 100baseT NIC will reduce the maximum number of users you can support.

The real challenge is not supporting thousands of users (IP Phones), it's connecting a sufficient number of PSTN connections to support those users.


Sincerely,
Brian LaVallee




On 12/18/13, 11:45 PM, bilal ghayyad wrote:

[mailinglist+asterisk a...]

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.

Once you get to the point where you start having hundreds of users then you really need to think about redundancy. OpenSIPS is a good way forward as it is very light weight and can easily handle tens of thousands of registered phones. As calls come in then it can forward them directly or if additional services are required such as possibly voicemail it can pass them onto one or more asterisk servers in a load balancing arrangement. Using a couple of database servers replicating to each other and a virtual IP address (or DNS SRV if the clients properly support it) will give you redundancy and allow you to add more server to expand the capacity.


On 19/12/13 05:19, Brian LaVallee wrote:

[rswagoner at gmail.com]

On Wed, Dec 18, 2013 at 9:45 AM, bilal ghayyad <bilmar_gh@yahoo.com (bilmar_gh@yahoo.com)> wrote:

[bilmar_gh at yahoo.com]

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


On Wednesday, December 18, 2013 6:10 PM, Tech Support <asterisk@voipbusiness.us> wrote:

#yiv9305735573 #yiv9305735573 -- _filtered #yiv9305735573 {font-family:Helvetica;panose-1:2 11 6 4 2 2 2 2 2 4;} _filtered #yiv9305735573 {font-family:Wingdings;panose-1:5 0 0 0 0 0 0 0 0 0;} _filtered #yiv9305735573 {panose-1:2 4 5 3 5 4 6 3 2 4;} _filtered #yiv9305735573 {font-family:Calibri;panose-1:2 15 5 2 2 2 4 3 2 4;} _filtered #yiv9305735573 {font-family:Tahoma;panose-1:2 11 6 4 3 5 4 4 2 4;} #yiv9305735573 #yiv9305735573 p.yiv9305735573MsoNormal, #yiv9305735573 li.yiv9305735573MsoNormal, #yiv9305735573 div.yiv9305735573MsoNormal {margin:0in;margin-bottom:.0001pt;font-size:12.0pt;} #yiv9305735573 a:link, #yiv9305735573 span.yiv9305735573MsoHyperlink {color:blue;text-decoration:underline;} #yiv9305735573 a:visited, #yiv9305735573 span.yiv9305735573MsoHyperlinkFollowed {color:purple;text-decoration:underline;} #yiv9305735573 span.yiv9305735573EmailStyle17 {color:#1F497D;} #yiv9305735573 .yiv9305735573MsoChpDefault {font-size:10.0pt;} _filtered #yiv9305735573 {margin:1.0in 1.0in 1.0in 1.0in;} #yiv9305735573 div.yiv9305735573WordSection1 {} #yiv9305735573 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 ghayyadSent: Wednesday, December 18, 2013 9:46 AMTo: Asterisk Users Mailing List - Non- Commercial DiscussionSubject: [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

[mailinglist+asterisk a...]

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.


On 20/12/13 10:08, bilal ghayyad wrote:

[webaccounts at jgoettg...]