Hi The problem got fixed. The solution was to use pulseaudio with the echo-cancel-module. Because of echo, the audio appeared like it was distorted and most of the times there was a howling sound. With the command line #pactl load-module echo-cancel-module and then running the pjsip app resolved the issue. Thanks & Regards Varma SVRP From: pjsip [mailto:pjsip-bounces@xxxxxxxxxxxxxxx] On Behalf Of Bill Gardner Sent: Tuesday, April 22, 2014 12:06 AM To: pjsip at lists.pjsip.org Subject: Re: PJSIP Audio fades, distorts and inaudible on ARM-i.MX53 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@xxxxxxxxxxxxxxx] On Behalf Of Bill Gardner Sent: Thursday, April 10, 2014 7:35 PM To: pjsip at lists.pjsip.org Subject: Re: 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@xxxxxxxxxxxxxxx [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 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 _______________________________________________ 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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