Hi, The best tool for debugging pjsip is the log file, you should make sure a log is created somewhere that you can access. EC creation is logged. I think EC is always created, even if speex is disabled in config file it will create pjsip echo suppressor. Identify which audio path is garbled. Mic -> RTP? RTP->speaker? You can capture RTP stream with wireshark. You can also capture any conference port entry to WAV file pretty easily. This help narrow down where the garbling is occurring. I would start by making sure audio playback and recording works on your device. Bill On 4/20/2014 11:47 PM, Varma wrote: > > Hi, > > I digged deep and found out that the packet loss was due to network > issue. I got that resolved. Now there is minimal packet loss of 1%-3% > occasionally. > > The audio is actually garbled. It still is. I found out that the > echo-cancellation was not kicking in. I put some debug statements in > the echo_create() function which would've printed on the console if EC > was kicking in. I am running the PJSUA app from the samples. How do I > tell the app to invoke echo-canceller forcibly? > > Thanks & Regards > > Varma SVRP > > *From:*pjsip [mailto:pjsip-bounces at lists.pjsip.org] *On Behalf Of > *Bill Gardner > *Sent:* Thursday, April 10, 2014 7:35 PM > *To:* pjsip at lists.pjsip.org > *Subject:* Re: [pjsip] PJSIP Audio fades, distorts and inaudible on > ARM-i.MX53 > > Hi, > > The packets stats show more loss than one would expect on a LAN, > enough to make it sound a bit garbled, but there's nothing to indicate > why your audio is fading away completely. You could try a wireshark > capture to verify audio is correctly sent from both endpoints, and > similarly you can set your ARM endpoint to automatically record to WAV > file. That should tell you something. Also check the log to see if you > are getting lots of master sound underflows, that would indicate CPU > is out of gas. Or maybe try audio only call without video. > > Regards, > > Bill > > On 4/10/2014 7:46 AM, Varma wrote: > > Hi, > > I ran the sndtest and capture the call dump. Does this say > anything about the audio going bad. > > # ./sndtest > > 08:13:14.760 sndtest.c !Found 2 devices: > > 08:13:14.761 sndtest.c 0: default:CARD=imx3stack > (capture=1, playback=1) > > 08:13:14.761 sndtest.c 1: sysdefault:CARD=imx3stack > (capture=1, playback=1) > > 08:13:15.220 sndtest.c Testing playback device > default:CARD=imx3stack > > 08:13:15.220 sndtest.c Testing capture device > default:CARD=imx3stack > > 08:13:15.424 sndtest.c Please wait while test is in > progress (~11 secs).. > > 08:13:26.581 sndtest.c Dumping results: > > 08:13:26.582 sndtest.c Parameters: clock rate=8000Hz, 80 > samples/frame > > 08:13:26.582 sndtest.c Playback stream report: > > 08:13:26.582 sndtest.c Duration: 9s.990 > > 08:13:26.582 sndtest.c Frame interval: min=0.029ms, > max=67.945ms > > 08:13:26.582 sndtest.c Jitter: min=9.948ms, avg=26.685ms, > max=67.913ms > > 08:13:26.583 sndtest.c Capture stream report: > > 08:13:26.583 sndtest.c Duration: 9s.980 > > 08:13:26.583 sndtest.c Frame interval: min=0.092ms, > max=63.500ms > > 08:13:26.583 sndtest.c Jitter: min=9.893ms, avg=26.536ms, > max=63.406ms > > 08:13:26.583 sndtest.c Checking for clock drifts: > > 08:13:26.583 sndtest.c Sound capture is 80 samples faster > than playback at the end of the test (average is 8 samples per second) > > 08:13:26.583 sndtest.c Test completed with some warnings > > ======================================================= > > Call statistics: > > [DISCONNCTD] To: > <sip:192.168.1.156>;tag=3d928773698d4280957744e929d8aa73 > > Call time: 00h:04m:42s, 1st res in 4587 ms, conn in 4642ms > > #0 audio PCMA @8kHz, sendrecv, peer=192.168.1.156:4004 > > SRTP status: Not active Crypto-suite: > > RX pt=8, last update:00h:00m:01.465s ago > > total 12.6Kpkt 2.03MB (2.53MB +IP hdr) > @avg=57.5Kbps/71.9Kbps > > pkt loss=249 (1.9%), discrd=0 (0.0%), dup=0 (0.0%), > reord=3 (0.0%) > > (msec) min avg max last dev > > loss period: 20.000 21.652 100.000 20.000 6.131 > > jitter : 0.250 8.685 29.000 7.000 3.389 > > TX pt=8, ptime=20, last update:00h:00m:01.318s ago > > total 11.8Kpkt 1.89MB (2.37MB +IP hdr) > @avg=53.7Kbps/67.2Kbps > > pkt loss=435 (3.5%), dup=0 (0.0%), reorder=0 (0.0%) > > (msec) min avg max last dev > > loss period: 20.000 164.151 1540.000 80.000 41.859 > > jitter : 0.000 14.582 21.875 12.125 4.515 > > RTT msec : 3.082 21.143 73.908 14.831 18.172 > > 00:34:55.055 pjsua_media.c ......Call 1: deinitializing media.. > > 00:34:55.056 pjsua_media.c ........Media stream call01:0 is destroyed > > 00:34:56.055 pjsua_aud.c Closing sound device after idle for 1 > second(s) > > 00:34:56.055 pjsua_app.c .Turning sound device OFF > > 00:34:56.055 pjsua_aud.c .Closing default:CARD=imx3stack sound > playback device and default:CARD=imx3stack sound capture device > > Thanks & Regards > > Varma SVRP > > *From:*pjsip-bounces at lists.pjsip.org > <mailto:pjsip-bounces at lists.pjsip.org> > [mailto:pjsip-bounces at lists.pjsip.org] *On Behalf Of *Varma > *Sent:* Monday, April 07, 2014 9:04 AM > *To:* pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org> > *Subject:* [pjsip] PJSIP Audio fades, distorts and inaudible on > ARM-i.MX53 > > Hi, > > I built PJSIP and run on Freescale ARM i.MX53. The video call > works well. The issue is with the audio. > > The call gets initiated from the i.MX base board to a mobile > running the csipsimple android app. The audio is heard properly on > the mobile but not on the i.MX board. The audio is heard clearly > on for 1-2 secs and it starts to fade out or distorted. Sometimes > audio is both distorted and faded. I tried all the steps in the > audio troubleshooting as mentioned in the PJSIP website. > > I checked my sound output by playing an audio file. It plays well. > > Following steps I tried. > > 1.Tried using PCMU and PCMA as the board supports only PCM. SPEEX > is disabled on the board. > > 2.Disabled the echo canceller. Did not have any effect on the result. > > 3.In-call volume to increased 5.0x, 10.0x and went up to 100.0x. > Every time I increased this the audio gets worser. > > 4.Tried forcing 8KHz sample. Still no improvement. > > Can you suggest what could fix my issue? > > I could not get the packet statistics. Somehow the TX and RX > packet count always shows zero when I use the 'dq' while in call. > > Thanks & Regards > > Varma SVRP > > Technical Lead | Orvito Technologies India Pvt Ltd. > > M: +91-9032867017 > > Description: Description: Description: Description: > cid:image002.png at 01CDD3CC.69D07270 > > 8-2-269/S/41 | Plot No.: 41 | Sagar Society| Banjara Hills | Road > No.: 2 | Hyderabad - 500038 > > www.orvito.com <http://www.orvito.com/> > > > > > _______________________________________________ > > Visit our blog:http://blog.pjsip.org > > > > pjsip mailing list > > pjsip at lists.pjsip.org <mailto: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... 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