[pjsip] Regarding Conference

Abdul Khadar abdul.khadar at pyronetworks.com
Mon May 26 06:21:16 EDT 2008


Hi ,
we are using jain-sip for sip stack. For handling call control. For 
Media purpose we are using pjsip.

For a single call it is working fine. when I started this library for 
conference , at that time voice is breaking.

Suse Linux Enterprise edition 10.
64-bit machine.
yamaha sound card.

I had tested with pjsua demo, it was working fine with different call 
conferences.

So , I developed Native library using C for Java. confsample demo code I 
have used to write my own library functions.

Calls are connecting properly and voice also distributed equally to all 
calls. but the problem I am facing is breaking voice.

please help me in this.

Benny Prijono wrote:
> On Fri, May 23, 2008 at 1:12 PM, Abdul Khadar
> <abdul.khadar at pyronetworks.com> wrote:
>> Hi,
>>
>> I have developed a shared library using pjsip.  By using this pjsip shared
>> library I want to create conference with a java program. For a single call
>> it is working fine , when I am placing more than one call and I am adding
>> this port to the conference pool.
>>
>> After call is  entered into conference voice reaching to all persons who are
>> in conference ,but voice is braking i.e. delay in voice.
>>
>> Below is the configuration for each port.
>>
>> Port #00:
>>   Name                    : /dev/dsp
>>   Sampling rate           : 16000 Hz
>>   Samples per frame       : 160
>>   Frame time              : 10 ms
>>   Signal level adjustment : tx=0, rx=0
>>   Current signal level    : tx=0, rx=0
>>   Transmitting to ports   : #1 #2
>>
>> Port #01:
>>   Name                    : strm0x530bf0
>>   Sampling rate           : 16000 Hz
>>   Samples per frame       : 160
>>   Frame time              : 10 ms
>>   Signal level adjustment : tx=0, rx=0
>>   Current signal level    : tx=0, rx=0
>>   Transmitting to ports   :  #0 #2
>>
>> Port #02:
>>   Name                    : strm0x531128
>>   Sampling rate           : 16000 Hz
>>   Samples per frame       : 160
>>   Frame time              : 10 ms
>>   Signal level adjustment : tx=0, rx=0
>>   Current signal level    : tx=0, rx=0
>>   Transmitting to ports   :  #0 #1
>>
>> this configuration changed to 8000 hz, 240 sample 20 ms also, when I have
>> this sample rates voice was more breaking, so I change to 10ms. It was
>> little bit ok but still it is breaking.
>>
>
> What platform are you running this on? Can it handle more than one calls?
> For troubleshooting audio problems please consult this page:
> http://trac.pjsip.org/repos/wiki/sound-problems
>
> Cheers
>  Benny
>
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