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.

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@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@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >

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@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@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > _______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list pjsip@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.

Since the cause of the distortion can vary, it is difficult to analyze the problem without any hints, for ex:source code of default_attr implementation, RTP packet capture, etc. There are already a lot of sample codes of codec wrappers in the pjmedia-codec :) Perhaps you need to make sure these things are correct: 1. Please make sure you have done the codec unit testing. 2. Recheck intensely every values set in the pjmedia_codec_param in default_attr() implementation. Regards, nanang On 13/03/2008, sre kdkjf <kk_kksri@yahoo.com> wrote: > 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@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@lists.pjsip.org > > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@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. > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >

On 3/13/08, Nanang Izzuddin <nanang.izzuddin@gmail.com> wrote: > Since the cause of the distortion can vary, it is difficult to analyze > the problem > without any hints, for ex:source code of default_attr implementation, > RTP packet capture, etc. > > There are already a lot of sample codes of codec wrappers in the > pjmedia-codec :) > Perhaps you need to make sure these things are correct: > 1. Please make sure you have done the codec unit testing. > 2. Recheck intensely every values set in the pjmedia_codec_param in > default_attr() implementation. And one more thing perhaps. Since G.723.1 has SID frame, make sure the decoder can parse the frames properly. -benny