30 simultaneus calls

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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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