[asterisk-users] Internal timing under load is critical ?

Steve Edwards asterisk.org at sedwards.com
Wed Jul 30 20:33:53 CDT 2014

Please don't top post.

Please keep the thread only on the list.

> On Thursday, July 31, 2014 12:16 AM, Steve Edwards 
> <asterisk.org at sedwards.com> wrote:
> I'm running Asterisk 11.11.0 on an HP ProLiant DL320e Gen8 E3-1240v2.
> 1,300 calls with no audio issues.

On Wed, 30 Jul 2014, babak wrote:

> 1300 calls include playback voices ?

The test scenario was for the first server to originate calls (via call 
files) to the second server and then 'playback()' a long file. The second 
server would answer the call and then 'playback()' a long file. Audio was 
flowing in each direction.

Bandwidth was observed using 'iftop' as being in the 70mb to 80mb range in 
each direction (if I remember correctly).

I placed calls from a handset to confirm audio quality.

> which timing module you are using: res_timing_timerfd.so or 
> res_timing_kqueue.so or res_timing_dahdi.sores_timing_pthread.so

I used res_timing_timerfd.so. I'm finally making the leap from 1.2 to the 
current decade :) I read somewhere that this was the timer to use and it 
seems to be working fine for me.

I don't think the cores got much over 20% to 30% busy.

Various failures were observed on the console from running out of file 
descriptors. This was on a stock CentOS 6.5 install with no tweaks to bump 
up the max file descriptors.

The client only asked for 500 simultaneous calls so no further testing was 

Thanks in advance,
Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 PST
Newline                                              Fax: +1-760-731-3000

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