Question from a newbie

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This is awesome news!, thanks man.

 

C. Savinovich

 

 

From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org]
On Behalf Of Paulo Rog?rio Panhoto
Sent: Tuesday, July 07, 2009 11:54 AM
To: pjsip list
Subject: Re: Question from a newbie

 

There is a .NET wrapper for PJSIP @
http://sipekphone.googlepages.com/pjsipwrapper

2009/7/7 C. Savinovich <c.savinovich at itntelecom.com>

Hello Everybody, I am very good with Asterisk and I have been in computer
telephony for many years, and I am tired of having to deal with the huge
costs of third party softphone libraries and I have decided to finally write
my very own softphone from scratch!... I mean, it is ok for anyone to charge
what they want for their work, don?t get me wrong
 I just believe that if I
am in the business of CTI, I ought to do this from scratch, since it does
not make business sense for me to offer a software package in which the
customer has to pay for 2 developers.

 

I will appreciate if you can please inform about the following:

 

1)      Is pjsip entirely in C??, isn?t there an API to use with maybe
Microsoft?s C# ??

2)      Is there anybody that maybe sell me OCX or DLL I can use ?, I am a
vb.net programmer.

3)      Being a vb.net programmer, what are my choices to use pjsip? (other
that learn GNU C)?

 

Thanks for your answers

C. Savinovich

 

From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org]
On Behalf Of Senthil
Sent: Tuesday, July 07, 2009 11:12 AM
To: pjsip list
Subject: Re: windows mobile underflow

 

Hi,

I'm using PJSIP 1.3.

Thanks,
- Senthil

Brad Radaker wrote: 

Also, what version of PJSIP are you guys using? I didn?t see that listed in
this thread anywhere.

 

Thanks,

Brad

 

 

  _____  

From: Srivatsan Deenadayalan [mailto:srivatsan@xxxxxxxxxxx] 
Sent: Tuesday, July 07, 2009 9:05 AM
To: pjsip list
Subject: Re: windows mobile underflow

 

Hi Senthil,

Change clock rate in config file from 16000 to 8000, hope it should work.
Which handset are you using?

Thanks,
Srivatsan D

Senthil wrote: 

Hi Attila,

I'm developing a windows desktop SIP phone application using PJSUA.  I'm
facing the the same problem as you mentioned in your mail (i.e. "Underflow
message in log and the remote party not able to hear clearly). I tried as
you stated in your previous mail but no success. Any suggestion to improve
the voice quality. 

Can you send me your modified config_site.h file; just want to double check
what I missed.

Any suggestion is greatly appreciated.

Thanks & Regards,
- Senthil

Attila Nyers wrote: 

Hi!

I tried to use some define from PJ_CONFIG_MAXIMUM_SPEED in config_site.h to
get better performace

#   define PJ_SCANNER_USE_BITWISE    0
#   undef PJ_OS_HAS_CHECK_STACK
#   define PJ_OS_HAS_CHECK_STACK    0
#   define PJ_LOG_MAX_LEVEL            4    
#   define PJ_ENABLE_EXTRA_CHECK    0
#   define PJ_DEBUG                    0
#   define PJSIP_SAFE_MODULE        0
#   define PJ_HAS_STRICMP_ALNUM        0
#   define PJ_HASH_USE_OWN_TOLOWER    1
#   define PJSIP_UNESCAPE_IN_PLACE    1

and this time I get these in log:

 03:29:05.978   Master/sound  Buffer size adjusted from 5440 to 5161
(eff_cnt=4400)
 03:29:06.115   Master/sound  Buffer size adjusted from 6121 to 5758
(eff_cnt=4400)
 03:29:06.227   Master/sound  Buffer size adjusted from 5758 to 5340
(eff_cnt=4400)
 03:29:06.318   Master/sound  Buffer size adjusted from 5340 to 4910
(eff_cnt=4400)
 03:29:06.442   Master/sound  Buffer size adjusted from 5230 to 5003
(eff_cnt=4400)
 03:29:06.579   Master/sound  Buffer size adjusted from 5323 to 5161
(eff_cnt=4400)
 03:29:06.692   Master/sound  Buffer size adjusted from 5161 to 4996
(eff_cnt=4400)
 03:29:06.805   Master/sound  Buffer size adjusted from 5956 to 5555
(eff_cnt=4400)
 03:29:06.910   Master/sound  Buffer size adjusted from 5555 to 5086
(eff_cnt=4400)
 03:29:07.205   Master/sound  Buffer size adjusted from 5406 to 4978
(eff_cnt=4400)




2009/7/7 Attila Nyers <atixlevlist at gmail.com>

Hi!

I can hear perfectly the other side, but my voice is metallic on the other
side. With PJSUA "desktop windows edition" there is no problem with sound

my CPU usage is on WinMo under 50% at all the time

// config_site.h
#define PJ_WIN32_WINCE 1
#include "config_site_sample.h"

// PJMEDIA
media_cfg.clock_rate = 8000;
media_cfg.audio_frame_ptime = 40;
media_cfg.ec_tail_len = 200;
media_cfg.max_media_ports = 8;
media_cfg.thread_cnt = 1;
media_cfg.ec_options = PJMEDIA_ECHO_SIMPLE;


part of the log:

22:50:21.019   strm0020B154  JB shrinking 1 frame(s), cur size=28
 22:50:21.081   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
 22:50:21.105   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
 22:50:21.153   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
 22:50:21.201   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
 22:50:21.225   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
 22:50:21.273   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
 22:50:21.434   strm0020B154  JB shrinking 1 frame(s), cur size=28
 22:50:21.466   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
 22:50:21.514   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
 22:50:21.562   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
 22:50:21.585   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
 22:50:21.633   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
 22:50:21.678   strm0020B154  JB shrinking 1 frame(s), cur size=28
 22:50:21.825   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
 22:50:21.873   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
 22:50:21.922   Master/sound  Underflow, buf_cnt=0, will generate 1 frame
 22:50:21.945   Master/sound  Underflow, buf_cnt=0, will generate 1 frame 

bye,
Attila 




-- 
?dv,
Attila



 
 



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_______________________________________________
Visit our blog: http://blog.pjsip.org
 
pjsip mailing list
 
pjsip at lists.pjsip.org
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Visit our blog: http://blog.pjsip.org

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