Hi Nanang yes i have added default_attr implementation of G723 codec, in my applicatoin. but the problem i am getting is only the distortion i.e. noise (lot of spike noise) is coming continuouly when we are in call. so i want to reduce that noise i.e distortion. how to avoid this. if you are able to send the sample code it will be great helpful for me. thankyou. Nanang Izzuddin <nanang.izzuddin at gmail.com> wrote: 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 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 > > _______________________________________________ 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 --------------------------------- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080312/ccd3a8c7/attachment.html
