Hi i did the same way how you told to me... But as the audio is not at all clear...it is coming lot of choppy..with lot of disturbance. i am sending the G723.c source file....how i made changes.. could you please go through this file and let me know what changes i need to make out. if it works fine it would be great help ful for me... i am waiting for your valuable reply. Thankyou. Nanang Izzuddin <nanang at pjsip.org> wrote: Sorry, this should be: bytes per frame: 320 -> 480 encoded bytes per frame: 33 -> 24 samples per frame: 160 -> 240 Regards, nanang 2008/6/6 Nanang Izzuddin : > Hi, > > It seems to be G.723.1 (G.723 is different codec). For bitrate > 6.3kbps, the attributes must be: > attr->info.avg_bps = 6400; // integer calculation in stream may cause > truncation, bigger value is safer, and it won't affect the real > bitrate. > attr->info.pcm_bits_per_sample = 16; // as it should be > (the rest is fine) > > Moreover, as you said your reference is GSM wrapper, there are > hardcoded values you need to modify in your G.723.1 wrapper: > bytes per frame: 320 -> 240 > encoded bytes per frame: 33 -> 24 > samples per frame: 160 -> 120 > > Good luck! > > nanang > > > 2008/6/6 sre kdkjf : >> >> Hi All, >> >> >> i am making a sample application using pjsip stack. in the pjsip stack i am trying to include audio codec G723. >> >> i had added G723 codec in the same way of GSM Codec, as it is added in the existing pjsip stack. >> whenever i made a call from A to B, once the call is in connected state, G723 rtp data is flowing from one end to other end. >> >> the problem is, the audio is choppy, in the rtp data G723 data is flowing from A to B. But the no.of frames flowing from one end to other end is coming less than 24. >> >> And the bitrate i am using here is 6.3kbps. even then the bitrate coming is 5.3 kbps. >> >> whenever i am using microphone to talk, the bit rate i am getting is 5.3 kbps. >> >> >> when the no.of frames are getting less than the actual frame rate i am getting error. >> >> PJMEDIA_CODEC_EFRMTOOSHORT >> >> so whenever the no. of frames per packet is less than the actual size, the output buffer is also less than the actual rate. >> >> at that time i am getting an error like >> >> PJMEDIA_CODEC_EPCMTOOSHORT. >> >> the attribute values i am using for G723 codec are as given below. >> >> 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; >> >> even i had changed the bits per samples to 16. even then the same problem i am getting. >> >> >> so any body please help me out, to how to solve the audio issues by using G723 codec. >> >> waiting for valuable reply. >> >> Thankyou. >> >> _______________________________________________ >> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080609/91494a6c/attachment-0001.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: g723.c Url: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080609/91494a6c/attachment-0001.c
