30 simultaneus calls

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/*
- use master clock (not sound device) for the clock source, since
sound device may be disturbed by high CPU load and CPU spikes.
- use clock rate 8000 Hz for the master clock (or conf bridge, if you
use it) to avoid resampling.

*/

Thanks very much for this complete answer.

I didn't understand what you mean with 'use master clock'.
I don't user any hardware but only your wav_player methods.
The clock rate is 8000 Hz by default, isn't it ?

Regards,

Frederic

 
Nanang Izzuddin a ?crit :
> Hi,
>
> G711 is very low complexity codec (see
> http://trac.pjsip.org/repos/wiki/PJMEDIA-MIPS, stream with G711 in P3
> 700MHz only cost < 0.1% CPU), so the codec shouldn't be the problem.
> Also your CPU utilization seems to be relatively low (although it
> supposed to be lower than 30%). So it's not the CPU limitation.
>
> Please see this, in case you haven't:
> https://trac.pjsip.org/repos/wiki/FAQ#high-perf.
> And also please try this, in case you haven't:
> - use master clock (not sound device) for the clock source, since
> sound device may be disturbed by high CPU load and CPU spikes.
> - use clock rate 8000 Hz for the master clock (or conf bridge, if you
> use it) to avoid resampling.
>
> Btw, is there any chance that the problem is in the caller/tester
> (e.g: the app tester can only handle 20 concurrent calls, so you
> should run 2 instances of app tester on 2 different machines)?
>
> Regards,
> nanang
>
>
> On 03/09/2008, Fr??d??ric CLEMENT <fclement at viatelecom.com> wrote:
>   
>> My Load average is about 1.00 (34 % CPU utilization).
>> The codec I am using is Alaw (g711).
>> Can I expect to handle 100 calls with a Core2Duo or QuadCore Xeon ?
>>
>> Should I user another codec ? My switch have many implemented. (Dialogic
>> Cantata IMG).
>>
>> Regards,
>>
>> Frederic
>>
>>
>> Nanang Izzuddin a ?crit :
>> Hi,
>>     
>
> Which codec are you using? How about the CPU utilization? Since
>   
>> it's
>>     
> may be caused by CPU limitation in doing media processing
>   
>> (especially
>>     
> encoding).
>
> Regards,
> nanang
>
>
> On 03/09/2008, Fr??d??ric CLEMENT
>   
>> <fclement at viatelecom.com> wrote:
>>     
>
>   
>> Hi again all,
>>     
>
> I am still building an IVR server and testing the max
>   
>> possible account of
>>     
> calls that can be handled by pjsip.
> I set
>   
>> PJSUA_MAX_CALLS to 100
>>     
> The value inf config_site.h for
>   
>> PJMEDIA_SOUND_BUFFER_COUNT is set to 16.
>>     
> My application just take calls and
>   
>> play a sound file looped. I use the pjsua
>>     
> high level methods
>   
>> (pjsua_player_create ....)
>>     
> The machine is a Pentium IV 1.9 Ghz with enough
>   
>> memory.
>>     
>
> When I send 20 calls, the sound stays being correct.
>
> If I send 30
>   
>> simultaneous calls, I get the same result than described on
>>     
> your site
>   
>> :
>>     
>
> http://trac.pjsip.org/repos/wiki/audio-problem-dropouts
>
> The
>   
>> sound I have is the same than the stutter.wav found
>> there.
>>     
>
> http://trac.pjsip.org/repos/attachment/wiki/audio-problem-dropouts/stutter.wav
>
> Wich
>   
>> optimisations can be done ? My goal is about 100 simultaneous
>> calls.
>>     
>
> Regards,
>
> Frederic
>
> (Sorry again for my very bad
>   
>> english)
>>     
>
> _______________________________________________
> Visit
>   
>> our blog: http://blog.pjsip.org
>>     
>
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>   
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>>     
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>
>
>
>   
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>>     
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>   
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>>     
>
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>
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