Thanks for the feedback. In this case SSD disks you think it solves?
Eduardo 2014-07-23 18:01 GMT-03:00 Ron Wheeler <[email protected]>: > 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 < > [email protected]> 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: [email protected] > 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
