G723 codec problem

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Since the cause of the distortion can vary, it is difficult to analyze
the problem
without any hints, for ex:source code of default_attr implementation,
RTP packet capture, etc.

There are already a lot of sample codes of codec wrappers in the
pjmedia-codec :)
Perhaps you need to make sure these things are correct:
1. Please make sure you have done the codec unit testing.
2. Recheck intensely every values set in the pjmedia_codec_param in
default_attr() implementation.

Regards,
nanang


On 13/03/2008, sre kdkjf <kk_kksri at yahoo.com> wrote:
> Hi Nanang
> yes i have added default_attr implementation of G723 codec, in my
> applicatoin.
>
> but the problem i am getting is only the distortion i.e. noise (lot of spike
> noise) is coming continuouly when we are in call. so i want to reduce that
> noise i.e distortion.
>
> how to avoid this.
> if you are able to send the sample code it will be great helpful for me.
>
> thankyou.
>
> Nanang Izzuddin <nanang.izzuddin at gmail.com> wrote:
> Hi Sre,
>
> Could you reveal your g723's pjmedia_codec_factory_op -> default_attr
> implementation?
> And also sending some RTP capture should be helpful.
>
> nanang
>
>
> On 12/03/2008, sre kdkjf wrote:
> > Hi All
> >
> > i am trying to include G723 Codec in pjsip stack.
> >
> > the way how GSM codec is included i had added G723 codec to PJSIP Stack.
> >
> > in rtp alos i had included G723 codec details.
> >
> > the problem i am getting is once after the call is connected thr rtp G723
> is
> > flowing from one end to other end.
> >
> > the only problem is the audio is getting some chopping. i.e. audio is
> coming
> > with lot of noise.
> >
> > please tell me how to avoid the noise when the call is in connected
> status.
> >
> > it would be great helpful if any body solves my problem.
> >
> > Thankyou.
> >
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