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] *On Behalf Of *Varma > *Sent:* Monday, April 07, 2014 9:04 AM > *To:* 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 > 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/20140410/b1f76e64/attachment.html> -------------- next part -------------- A non-text attachment was scrubbed... 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