G723.1 codec Audio problem in pjsip

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Hi,

Sorry for the late response. The wrapper looks fine except for these parts:
- since you are using older version, try to set attr->info.avg_bps to 6400
or use the latest release 0.9.0, there are some updates related to codec
wrapper.
- try with both VADs disabled first (it seems the codec implementation has
VAD and it is enabled, there is also pjmedia VAD/silence det which actually
unnecessary for codec impl that has built in VAD). This VAD things seems
to cause problem in the codec parsing (hardcoded to accept only payload
which has size multiplication of 24 bytes).

I haven't tried it my self, but I would say that I couldn't see anything
that may cause G723.1 doesn't work with pjsip stack. And fyi, there is also
working codec that uses ptime=30, i.e: iLBC.

Regards,
nanang


2008/6/26 sre kdkjf <kk_kksri at yahoo.com>:

>   Hi All,
>
>
>
> i am tryting to make a sample application using pjsip stack, the audio
> codec i am using is G723.1 Codec.
>
>
>
>
>
> The audio is getting choppy, its not at all clear after the call is in
> connected state,
>
>
>
>
>
> i had changed so many parameters in the class of G723.c which i wrote.
>
>
>
> i want to know weather pj sip stack supports G723.1 Audio codec or not.
>
>
>
> some times i am getting error as
>
> PJMEDIA_CODEC_EPCMTOOSHORT
>
>
>
> this error i am getting only when the input signal size is less than 24
>
>
>
> how this problem can be solved.?
>
>
>
> As i am sending the file as an attachment.
>
>
>
>
>
> Any body please check that and let me know where the problem is.
>
>
>
> i want to know anything i want to change in the pj media to support G723.1
> codec.
>
>
>
> As the rtp flow is happening at both ends, but the audio is not clear its
> with lot of choppy, and lot of noise is coming.
>
>
>
>
>
> any body please help me out to solve this problem.
>
>
>
> Thankyou.
>
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip at lists.pjsip.org
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>
>
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