Please change: attr->info.pcm_bits_per_sample = 24; with attr->info.pcm_bits_per_sample = 16; and please make sure the active bitrate used by your g723.1 codec implementation is really 6.3 kbps. Regards, nanang On 17/03/2008, sre kdkjf <kk_kksri at yahoo.com> wrote: > 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. > > ________________________________ > Never miss a thing. Make Yahoo your homepage. > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >
