Hi All
i am trying to include G723 codec in PJSIP Stack.
i had included G723 codec details in the stack, and in rtp also.
the problem i am getting is when i am in call, in the rtp also G723.1 is flowing from one end to end. But there is lot of noise disturbence. i.e. lot of distortion is coming.so that both users not able to hear the audio properly.
the default attribute which i have included for G723 codec is as given below.
PJ_UNUSED_ARG(factory);
PJ_UNUSED_ARG(id);
pj_bzero(attr, sizeof(pjmedia_codec_param));
attr->info.clock_rate = 8000;
attr->info.channel_cnt = 1;
attr->info.avg_bps = 6300;
attr->info.pcm_bits_per_sample = 24;
attr->info.frm_ptime = 30;
attr->info.pt = PJMEDIA_RTP_PT_G723;
attr->setting.frm_per_pkt = 1;
attr->setting.vad = 1;
#if !PLC_DISABLED
attr->setting.plc = 1;
#endif
so is there any changes i have to make in the source.
so can you please tell me how to reduce the distortion in audio i.e. how to reduce noise in audio, so that both parties can communicate can hear audio clearly.
if any body solved this problem it would be great helpful for me.
Thankyou.
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