30 simultaneus calls

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HI,

Thanks very much.
I will test this tomorrow.

Regards,

Frederic

On Thu, 4 Sep 2008 01:59:08 +0700, "Nanang Izzuddin" <nanang at pjsip.org>
wrote:
> Hi,
> 
> Sorry, it should be master port, one of pjmedia clock generator, here
> is the doc:
> http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__MASTER__PORT.htm
> 
> There is no default clock rate for wav player, wav player clock rate
> follows the wav file clock rate. So if you are using G711, please use
> 8000 Hz wav file to avoid resampling.
> 
> Btw, replacing wav player with mem player may increase the
> performance, not sure though.
> 
> Regards,
> nanang
> 
> 
> On 04/09/2008, Fr??d??ric CLEMENT <fclement at viatelecom.com> wrote:
>> /*
> - 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)
> 
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