Hi Benny/Nanang,
? I integrated G723 Codec in to PJSIP successfully,
These are the parameters iam using while configuring G723 to PJSIP
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 = 16;
??? attr->info.frm_ptime = 30;
??? attr->info.pt = PJMEDIA_RTP_PT_G723;
??? attr->setting.frm_per_pkt = 1;
??? attr->setting.vad = 1;
#if !PLC_DISABLED
??? attr->setting.plc = 1;
#endif
/* Enable high pass filter */
??? g723_set_hp(1);
??? /* Enable post filter */
??? g723_set_pf(1);
??? /* Enable VAD */
??? g723_set_vx(0);
???
??? /* Use 6.3kbps rate */
??? g723_set_rate(0);
In X SIP Server i created two Accounts A and B
If i call from A to B voice is audible at both sides with out disturbance.
while iam calling from A to PSTN Mobile Call,from A side voice is audible clearly at Mobile side voice with disturbance.
At the same time after call connected iam getting Bad RTP
?pjsua_app.c? Media for call 1 is active
?sound_port.c? EC activated
?strm00D158BC? VAD re-enabled
?strm00D158BC? Bad RTP pt 19 (expecting 4)
?strm00D158BC? Bad RTP pt 19 (expecting 4)
?strm00D158BC? Bad RTP pt 19 (expecting 4)
?strm00D158BC? Bad RTP pt 19 (expecting 4)
?strm00D158BC? Bad RTP pt 19 (expecting 4)
if i use the GSM codec wt PJSIP is provided,after call connecting everything fine.
wt is the problem while adding the G723 codec,could you explain clearly...
Regards
Rams
??
The INTERNET now has a personality. YOURS! See your Yahoo! Homepage. http://in.yahoo.com/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100312/78aaa624/attachment.html>
