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.
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