Hi Benny, I'm using MDA (no APS). The log file I usually open with MS Word may be that the reason you don't have the expected results. The test was on Nokia E71 connected to my wireless provider Access Point (Rogers Canada) calling a PSTN local number. Thanks, George. _____ From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org] On Behalf Of Benny Prijono Sent: Tuesday, May 12, 2009 10:08 AM To: pjsip list Subject: Re: audio problem with GSM on Symbian 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 <http://trac.pjsip.org/repos/wiki/audio-problem-breakups> 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 <mailto: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 <mailto: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 <http://trac.pjsip.org/repos/wiki/audio-problem-dropouts> drop-outs or "stutters": 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 <http://trac.pjsip.org/repos/wiki/audio-check-cpu> that CPU Utilization is not Too High, 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/20090512/019396b4/attachment-0001.html>
