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