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 <edua...@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 <sgriepent...@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 < >> edua...@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 <rwhee...@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 < >>>> edua...@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: rwhee...@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 >> > >
-- _____________________________________________________________________ -- 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