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@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org] On Behalf Of Bugra Cakir Sent: den 25 april 2010 10:03 To: pjsip list Subject: Re: 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 <mailto:sip%3Atest7 at telekom.com.tr> >;tag=xvy7KGP82TYBfYWJlAAMobbbHClKLJOx From: <sip:0312481xxxx@xxxxxxxxxxx <mailto: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 <http://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 <mailto:sip%3A0312481xxxx at telekom.com> >;tag=2055258867 To: <sip:test7 at telekom.com <mailto: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 <mailto: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 <http://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 <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/20100425/43aa9852/attachment-0001.html>
