Hi, Dont use ilbc please use pcma or pcmu and you have to compile with MINIMAL settings in config_site. http://svn.pjsip.org/repos/pjproject/branches/projects/iphone/pjlib/include/pj/config_site_sample.h You can find more information in README file for compiling Best Regards M.Ali VARDAR 2010/8/17 R?gis Montoya <r3gis.3r at gmail.com> > Sorry to update this topic but I think that I have some new clues on this > issue. > > (As reminder I'm the developer of the port of pjsip for android). > I have made a build with ilbc enabled. And I get the same logs that the one > described on this thread. It's also on arm architecture but devices have a > correct CPU (1Ghz). > The result is that if I use ilbc with mode 20ms, things are really good, > but if set mode to 30ms get exactly the same buffer resizing logs and sound > is really bad. > Maybe my audio driver is not well designed but don't think so since works > well with all other codecs (and when ilbc mode is 20). > Is there some constraints about buffers size/chunk read size on the audio > driver? > > Regis > > > 2010/4/25 Olle Frimanson <olle.frimanson at keystream.se> > >> Hi Bugra, >> >> >> >> We have also used pjsip on several different ARM?s so that is not the >> issue, what might be worth checking is what samplings rates your codec >> (AD/DA) supports. If it?s doesn?t supports 8000 resampling will be done in >> ALSA driver I think and this could be a problem. >> >> >> >> Also have you tried the basic stuff like connect a local call in PJSUA, >> play out a file and record from file does that work? >> >> >> >> BR/Olle >> >> >> ------------------------------ >> >> *From:* pjsip-bounces at lists.pjsip.org [mailto: >> pjsip-bounces at lists.pjsip.org] *On Behalf Of *Bugra Cakir >> *Sent:* den 25 april 2010 10:03 >> *To:* pjsip list >> *Subject:* Re: [pjsip] Buffer problem >> >> >> >> No way out with PCMU and clockrate ! Shoulda try something else. >> >> >> >> ./pjsua-armv7l-unknown-linux-gnu --username=test7 --password=123 >> --proxy=sip:85.123.66.44 --registrar sip:telekom.com --id >> sip:test7 at turktelekom.com.tr --add-codec pcmu --dis-codec speex >> --dis-codec ilbc --dis-codec GSM --dis-codec G722 --clock-rate=8000 >> --snd-clock-rate=8000 --capture-dev=1 --playback-dev=0 >> >> >> >> >> >> 19:20:06.402 pjsua_app.c Call 0 is DISCONNECTED [reason=200 (Normal >> call clearing)] >> >> 19:20:06.403 pjsua_app.c >> >> [DISCONNCTD] To: sip:03124818246 at telekom.com;tag=632633610 >> >> Call time: 00h:00m:28s, 1st res in 6678 ms, conn in 10920ms >> >> #0 PCMU @8KHz, sendrecv, peer=78.223.11.23:12240 >> >> RX pt=0, stat last update: 00h:00m:15.501s ago >> >> total 1.4Kpkt 230.2KB (288.0KB +IP hdr) @avg=57.0Kbps/71.3Kbps >> >> pkt loss=6 (0.4%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%) >> >> (msec) min avg max last dev >> >> loss period: 20.000 20.000 20.000 20.000 0.000 >> >> jitter : 0.000 0.313 11.625 0.125 0.824 >> >> TX pt=0, ptime=20ms, stat last update: 00h:00m:00.150s ago >> >> total 364pkt 58.2KB (72.8KB +IP hdr) @avg 14.4Kbps/18.0Kbps >> >> pkt loss=7 (1.9%), dup=0 (0.0%), reorder=0 (0.0%) >> >> (msec) min avg max last dev >> >> loss period: 20.000 46.667 100.000 20.000 32.731 >> >> jitter : 50.250 70.161 87.125 87.125 10.734 >> >> RTT msec : 12.054 15.701 17.456 12.054 2.131 >> >> 19:20:06.404 pjsua_media.c Media session for call 0 is destroyed >> >> 19:20:06.467 ec0x165cf8 Buffer size adjusted from 7194 to 6025 >> (eff_cnt=5512) >> >> 19:20:06.669 ec0x165cf8 Buffer size adjusted from 7208 to 5784 >> (eff_cnt=5512) >> >> 19:20:06.881 ec0x165cf8 Buffer size adjusted from 7527 to 6098 >> (eff_cnt=5512) >> >> 19:20:07.037 ec0x165cf8 Buffer size adjusted from 7310 to 5871 >> (eff_cnt=5512) >> >> >> >> >> >> On Apr 24, 2010, at 9:44 AM, P.Muge Ersoy wrote: >> >> >> >> Hi Bugra; >> >> I dont think it is a buffer problem.. >> I compiled pjsip on arm processor and it was working just fine.. Here what >> i did.. >> I compiled it with PJ_CONFIG_MINIMAL_SIZE at config_site.h.. >> >> I ve never used iLBC codec.. coz it is too heavy for the processor.. >> instead use G.711.. you will hear more proper voice.. >> >> Selamlar Muge :) >> >> >> >> On Sat, Apr 24, 2010 at 9:09 AM, Bugra Cakir <bugra.cakir at argela.com.tr> >> wrote: >> >> Hi, >> >> I'm running pjsip-1.6 latest trunk on an beagle board based on ARM >> processor. >> While i'm running pjsua agent it is complaining about >> >> >> 17:22:12.483 ec0x165cf8 Buffer size adjusted from 1922 to 1443 >> (eff_cnt=1440) >> 17:22:12.609 ec0x165cf8 Buffer size adjusted from 2083 to 1604 >> (eff_cnt=1440) >> 17:22:12.643 ec0x165cf8 Buffer size adjusted from 1924 to 1445 >> (eff_cnt=1440) >> 17:22:12.843 ec0x165cf8 Buffer size adjusted from 1765 to 1286 >> (eff_cnt=1440) >> 17:22:14.623 ec0x165cf8 Buffer size adjusted from 2246 to 1767 >> (eff_cnt=1320) >> 17:22:14.643 ec0x165cf8 Buffer size adjusted from 1767 to 1288 >> (eff_cnt=1320) >> 17:22:15.663 ec0x165cf8 Buffer size adjusted from 1608 to 1129 >> (eff_cnt=1230) >> >> >> and after i initiate a call during 180 ringing and after 200 ok >> >> >> 17:24:04.540 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:04.689 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:04.740 ec0x165cf8 Buffer size adjusted from 1771 to 1292 >> (eff_cnt=1198) >> 17:24:04.814 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:04.865 ec0x165cf8 Buffer size adjusted from 1612 to 1133 >> (eff_cnt=1198) >> 17:24:04.917 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:05.046 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:05.070 ec0x165cf8 Buffer size adjusted from 1773 to 1294 >> (eff_cnt=1198) >> 17:24:05.157 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:05.186 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:05.259 ec0x165cf8 Buffer size adjusted from 1614 to 1135 >> (eff_cnt=1198) >> 17:24:05.321 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:05.449 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:05.526 ec0x165cf8 Buffer size adjusted from 2095 to 1616 >> (eff_cnt=1198) >> 17:24:05.559 ec0x165cf8 Buffer size adjusted from 1616 to 1137 >> (eff_cnt=1198) >> 17:24:05.575 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:05.694 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:05.759 ec0x165cf8 Buffer size adjusted from 1777 to 1298 >> (eff_cnt=1198) >> 17:24:05.834 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:05.880 ec0x165cf8 Buffer size adjusted from 1618 to 1139 >> (eff_cnt=1198) >> 17:24:05.947 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:06.064 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:06.227 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:06.299 ec0x165cf8 Buffer size adjusted from 2099 to 1620 >> (eff_cnt=1198) >> 17:24:06.322 ec0x165cf8 Buffer size adjusted from 1620 to 1141 >> (eff_cnt=1198) >> 17:24:06.409 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:06.436 ec0x165cf8 Buffer size adjusted from 1461 to 982 >> (eff_cnt=1138) >> 17:24:06.533 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:06.667 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:06.716 ec0x165cf8 Buffer size adjusted from 1622 to 1143 >> (eff_cnt=1138) >> 17:24:06.787 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:06.891 Master/sound Underflow, buf_cnt=0, will generate 1 frame >> 17:24:06.950 sound_port.c EC suspended because of inactivity >> 17:24:06.954 ec0x165cf8 Buffer size adjusted from 1783 to 1304 >> (eff_cnt=1138) >> 17:24:09.474 Master/sound Buffer size adjusted from 1600 to 1300 >> (eff_cnt=1273) >> 17:24:10.403 pjsua_core.c RX 423 bytes Request msg BYE/cseq=27568 >> (rdata0x176f5c) from UDP 2xx.1xx.1xx.xxx:5060: >> BYE sip:test7 at 94.54.xx.xx:5060;transport=UDP SIP/2.0 >> To: <sip:test7 at telekom.com.tr <sip%3Atest7 at telekom.com.tr> >> >;tag=xvy7KGP82TYBfYWJlAAMobbbHClKLJOx >> From: <sip:0312481xxxx@xxxxxxxxxxx <sip%3A0312481xxxx at telekom.com> >> >;tag=D2055258867AF >> Via: SIP/2.0/UDP 212.174.177.33:5060 >> ;branch=z9hG4bK-d87543-297e466706249ce1-1--d87543-;rport >> Call-ID: YBw7zbEzXBcctw-kYRH10mwveLf-WYsD >> CSeq: 27568 BYE >> Contact: <sip:212.174.177.33:5060> >> Max-Forwards: 69 >> Content-Length: 0 >> >> >> --end msg-- >> 17:24:10.404 pjsua_core.c TX 358 bytes Response msg 200/BYE/cseq=27568 >> (tdta0x1c3058) to UDP 2xx.1xx.1xx.xxx:5060: >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 212.174.177.33:5060 >> ;rport=5060;received=212.174.177.33;branch=z9hG4bK-d87543-297e466706249ce1-1--d87543- >> Call-ID: YBw7zbEzXBcctw-kYRH10mwveLf-WYsD >> From: <sip:0312481xxxx@xxxxxxxxxxx <sip%3A0312481xxxx at telekom.com> >> >;tag=2055258867 >> To: <sip:test7 at telekom.com <sip%3Atest7 at telekom.com> >> >;tag=xvy7KGP82TYBfYWJlAAMobbbHClKLJOx >> CSeq: 27568 BYE >> Content-Length: 0 >> >> >> --end msg-- >> 17:24:10.405 pjsua_app.c Call 1 is DISCONNECTED [reason=200 (Normal >> call clearing)] >> 17:24:10.406 pjsua_app.c >> [DISCONNCTD] To: sip:0312481xxxxx at telekom.com<sip%3A0312481xxxxx at telekom.com> >> ;tag=2055258867 >> Call time: 00h:00m:22s, 1st res in 5599 ms, conn in 12939ms >> #0 iLBC @8KHz, sendrecv, peer=212.174.177.62:12060 >> RX pt=113, stat last update: 00h:00m:02.478s ago >> total 264pkt 13.1KB (23.7KB +IP hdr) @avg=4.5Kbps/8.2Kbps >> pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%) >> (msec) min avg max last dev >> loss period: 0.000 0.000 0.000 0.000 0.000 >> jitter : 0.000 3.042 18.250 1.250 4.373 >> TX pt=100, ptime=30ms, stat last update: 00h:00m:15.268s ago >> total 356pkt 17.8KB (32.0KB +IP hdr) @avg 6.2Kbps/11.1Kbps >> pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) >> (msec) min avg max last dev >> loss period: 0.000 0.000 0.000 0.000 0.000 >> jitter : 26.000 39.125 52.250 52.250 13.125 >> RTT msec : 28.030 32.760 37.490 37.490 4.730 >> >> >> I'm hearing some part of voice activity from A and B party but they are >> not accurate. >> Is the problem related with media infrastructure or the platform it is >> running on ? >> >> Thank you >> >> >> >> >> >> >> >> _______________________________________________ >> 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 >> >> >> >> _______________________________________________ >> 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 > > -- -------------------- Sayg?lar?mla M.Ali VARDAR -------------- next part -------------- An HTML attachment was scrubbed... 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