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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