30 simultaneus calls

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Hi Nanang,

As I am a very new user of your library, I only use the high-level API 
(pjsua).
Would you have some sample code using the methods you describe ?

I can see following in the logs :

 06:42:44.886     resample.c  resample created: high qualiy, large 
filter, in/out rate=8000/16000
 06:42:44.886     resample.c  resample created: high qualiy, large 
filter, in/out rate=16000/8000

My Wav file has following format :

Input Filename : Acc.wav
Sample Size    : 16-bits
Sample Encoding: signed (2's complement)
Channels       : 1
Sample Rate    : 8000

What does it mean ?

Regards,

Frederic

CLEMENT Fr?d?ric a ?crit :
> 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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>>>       
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>>
>>
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>
>
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