[asterisk-users] Limit Asterisk

Thu Jul 24 07:05:51 CDT 2014

Thank you all for the answers. I will do tests to find the problem.

One other question I have, in the scenario that I sent, how bad would be to
transcode G711 to G729 in 70% of calls? There is a study that shows a
statistically loss of performance (concurrent calls) with active transcode?

tks

2014-07-24 8:54 GMT-03:00 Scott Griepentrog <sgriepentrog at digium.com>:

> Whether SSD drives allow you to add any additional calls depends entirely
> on whether or not they can be written to faster than the SAS drives you
> have.  My experience shows SSD's can be twice as fast as run-of-the-mill
> SATA, but the performance difference compared to SAS is likely not as
> great, and could even be worse.  You'll need to test two drives to find
> out.  I recommend mounting both to test them and copying a very large ISO
> file using dd which will give you the transfer rate when finished.  Then
> you should have your answer.
>
>
> On Wed, Jul 23, 2014 at 4:03 PM, Eduardo Leones <
> eduardo at ypytecnologia.com.br> wrote:
>
>> Thanks for the feedback.
>>
>> In this case SSD disks you think it solves?
>>
>>
>> Eduardo
>>
>>
>> 2014-07-23 18:01 GMT-03:00 Ron Wheeler <rwheeler at artifact-software.com>:
>>
>>  I would also do some math on the bandwidth requirement.
>>>
>>> If you divide your disk bandwidth by your recording bit rate what is the
>>> theoretical maximum number of calls that you can record at once? Assumes
>>> that you have infinite CPU and memory and that you can actually drive the
>>> disks at their maximum.
>>> If this comes out to 300, you are already there. If it comes out to
>>> 3000, you have something wrong in your setup or your assumptions and a
>>> target to work towards.
>>>
>>> What quality are you using in the recording? 44k per second(CD quality
>>> sound)  uses a lot more bandwidth than 3K (telephone quality)
>>> What encoding are you using?
>>> How low a bit rate can you use and still have usable recordings? If they
>>> are for legal or audit use, you can go pretty low. If you are recording
>>> soundtracks for reuse in training or publication, you may require higher
>>> bit rates.
>>>
>>> If you disable recording, how many simultaneous calls can you support?
>>> Just to be sure that recording is the issue.
>>>
>>> Ron
>>>
>>>
>>> On 23/07/2014 4:29 PM, Scott Griepentrog wrote:
>>>
>>>  Your bottleneck is most likely your drive bandwidth.  Even with SAS
>>> drives, you'll need to move to a raid 5+ solution with 6+ drives to
>>> continue to increase the concurrent calls, or use a storage appliance.
>>>
>>>  To confirm this, install the tool nmon and use the v and d options to
>>> bring up the resource usage indicators and drive busy/throughput statistics.
>>>
>>>
>>>
>>> On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones <
>>> eduardo at ypytecnologia.com.br> wrote:
>>>
>>>>  people
>>>>
>>>>  I have a running Asterisk 1.8.28 in great Dell server with two xeon
>>>> processors and 16gb of ram and HD SAS 15k (Raid 1). This server is
>>>> recording all calls (placed to record the audio in a ram disk), the entire
>>>> CDR goes straight to MySQL by cdr_mysql.so. Each call runs some validation
>>>> and AGI's have an auto dialer system that generates calls over the manager.
>>>> Calls originate and terminate via SIP (no transcode).
>>>>
>>>>  With this structure, even being a great server, we can not spend 150
>>>> simultaneous calls. When it reaches 140, the load average goes up a lot and
>>>> the calls start to get very bad audio, tear, etc.. Using the top we see
>>>> that all the processing is for asterisk. In this scenario, I think there is
>>>> some limitation in Asterisk, or even the manager due to the auto dialer.
>>>>
>>>>  Can anyone give me any tips where I can look where is the bottleneck?
>>>> I need to get at least 250 calls that server quality.
>>>>
>>>>  tks
>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
>>>> -- 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
>>>>
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>>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>>
>>>
>>>  --
>>>  [image: Digium logo]
>>> Scott Griepentrog
>>> Digium, Inc · Software Developer
>>> 445 Jan Davis Drive NW · Huntsville, AL 35806 · US
>>> direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029
>>> Check us out at: http://digium.com · http://asterisk.org
>>>
>>>
>>>
>>>
>>> --
>>> Ron Wheeler
>>> President
>>> Artifact Software Inc
>>> email: rwheeler at artifact-software.com
>>> skype: ronaldmwheeler
>>> phone: 866-970-2435, ext 102
>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- 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
>>
>
>
>
> --
> [image: Digium logo]
> Scott Griepentrog
> Digium, Inc · Software Developer
> 445 Jan Davis Drive NW · Huntsville, AL 35806 · US
> direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029
> Check us out at: http://digium.com · http://asterisk.org
>
> --
> _____________________________________________________________________
> -- 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
>
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