Hi Sre, Could you reveal your g723's pjmedia_codec_factory_op -> default_attr implementation? And also sending some RTP capture should be helpful. nanang On 12/03/2008, sre kdkjf <kk_kksri at yahoo.com> wrote: > 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. > > ________________________________ > Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it > now. > > > _______________________________________________ > 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 > >
