Hi George, First of all, are you using MDA or APS? Though we tested both before 1.2 release and they seem to be fine. And secondly, to localize the problem, what if you call another pjsua on the same LAN first and see how it goes. cheers Benny PS: the log seems to strip most of the newlines so can't read too much info there. On Thu, May 7, 2009 at 7:43 PM, George Evi <george.evi at ctcinc.ca> wrote: > Hi, > > > > I?m trying to use GSM/8000 codec with Pjsip library and have some audio > problems. Tests are done on a Nokia phone E71. > > I passed through the following principal points (others which are not > present some of them are not applicable): > > > > 2. Audio breakups<http://trac.pjsip.org/repos/wiki/audio-problem-breakups> > : > > > > 1. It's always recommended to check whether the > problem exists with the latest SVN version of the libraries, Yes (revision > 2668). > > > > 2. Check that CPU utilization is not too high, Yes > (can see a CPU load 19.47%, memory usage, function calls in profiler file). > > > > 3. Check for high network jitter, packet loss, Yes > (below is the statistical dump): > > > > 1. Pjsua Statistics, 5may09, 17h12: > > > > 1.1 GSM statistics (): > > > > --end msg--State changed from Null to Calling, event=TX_MSGTransaction > tsx0x749e5c state changed to Calling > > [CONFIRMED] To: sip:5145294251 at sip6.van.netvoice.ca<sip%3A5145294251 at sip6.van.netvoice.ca> > ;tag=as16e1281f > > Call time: 00h:00m:59s, 1st res in 2211 ms, conn in 21845ms > > SRTP status: Not active Crypto-suite: (null) > > #0 GSM @8KHz, sendrecv, peer=64.34.49.82:14512 > > RX pt=3, stat last update: 00h:00m:00.525s ago > > total 3.7Kpkt 125.3KB (277.1KB +IP hdr) @avg=12.5Kbps/27.8Kbps > > pkt loss=171 (4.3%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 > (0.0%) > > (msec) min avg max last dev > > loss period: 680.000 1,140.000 1,900.000 680.000 62.227 > > jitter : 0.250 12.555 4,135.000 0.250 8.533 > > TX pt=3, ptime=20ms, stat last update: 00h:00m:04.596s ago > > total 3.1Kpkt 103.1KB (228.4KB +IP hdr) @avg 10.3Kbps/22.9Kbps > > 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 : 41.875 112.100 206.875 75.625 10.972 > > RTT msec : 152.000 223.800 540.000 152.000 17.061 > > Jitter buffer empty (prefetch=19)Starting talk burst (level=58 > threshold=50)Start talksprut..Processing incoming message: Response msg > 200/BYE/cseq=21287 (rdata0x71e46c)RX 507 bytes Response msg > 200/BYE/cseq=21287 (rdata0x71e46c) from UDP 64.34.49.82:5060: > > SIP/2.0 200 OK > > > > - below are the statistics after using the PCMU codec: > > > > 1.2 PCMU statistics: > > > > --end msg--State changed from Null to Calling, event=TX_MSGTransaction > tsx0x74a09c state changed to Calling > > [CONFIRMED] To: sip:5145294251 at sip6.van.netvoice.ca<sip%3A5145294251 at sip6.van.netvoice.ca> > ;tag=as013a5bcb > > Call time: 00h:00m:39s, 1st res in 2484 ms, conn in 24565ms > > SRTP status: Not active Crypto-suite: (null) > > #0 PCMU @8KHz, sendrecv, peer=64.34.49.82:18934 > > RX pt=0, stat last update: 00h:00m:03.806s ago > > total 3.0Kpkt 491.8KB (614.8KB +IP hdr) @avg=63.7Kbps/79.6Kbps > > pkt loss=0 (0.0%), discrd=1 (0.0%), dup=0 (0.0%), reord=1 (0.0%) > > (msec) min avg max last dev > > loss period: 0.000 0.000 0.000 0.000 0.000 > > jitter : - 0.001 6.563 598.000 11.250 6.433 > > TX pt=0, ptime=20ms, stat last update: 00h:00m:01.676s ago > > total 2.6Kpkt 428.3KB (535.4KB +IP hdr) @avg 55.5Kbps/69.3Kbps > > 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 : 64.250 83.802 95.125 86.750 8.723 > > RTT msec : 173.000 209.417 283.000 197.000 64.301 > > Processing incoming message: Response msg 200/BYE/cseq=7437 > (rdata0x71e46c)RX 506 bytes Response msg 200/BYE/cseq=7437 (rdata0x71e46c) > from UDP 64.34.49.82:5060: > > SIP/2.0 200 OK > > Via: SIP/2.0/UDP > > > > 4. Check if audio is breaking up when playing file locally, Yes (run > ?sndtest.c?). > > > > 3. Audio drop-outs or "stutters"<http://trac.pjsip.org/repos/wiki/audio-problem-dropouts>: > > > > > 1. Checking that Microphone and Speaker are > Functioning Properly, N/A on the phone. > > > > 2. Check that there is no dangling call in the PBX<http://trac.pjsip.org/repos/wiki/audio-check-dangling-pbx-call>, > N/A on the phone. > > > > 3. Checking for Network Impairments of Incoming RTP > Packets, done in previous section. > > > > 4. Check that CPU Utilization is not Too High<http://trac.pjsip.org/repos/wiki/audio-check-cpu>, > already check in the previous section. > > > > 5. Try to enlarge PJMEDIA_SOUND_BUFFER_COUNT, no > need for the Pjsip version greater then 0.9. > > > > 6. Check for audio underflow/overflow, Yes (found jn the log file > traces of ?underflow? situation, see attached ?pjsua_MyE71_GSM_02.log? log > file). > > > > The result of test: > > > > - caller side: when listen welcome message voice is with no > breakups, no stuttering. > > > > - callee (remote ) side: breakups, stuttering & echo present, > bad quality. > > > > - I have also a WAV file example sent to Benny. > > > > Thanks, > > George. > > > > _______________________________________________ > 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/20090512/78bb54fb/attachment.html>
