[asterisk-users] Limit Asterisk

Thu Jul 24 12:07:06 CDT 2014

Another question, what audio format I use in MixMonitor to maintain a
connection with reasonable quality and reduce the use of I / O disk? Today
I use wav.

tks

2014-07-24 9:05 GMT-03:00 Eduardo Leones <eduardo at ypytecnologia.com.br>:

> 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
>>>>>
>>>>> 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
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> 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
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140724/87e21764/attachment.html>