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.
-
Tried using PCMU and PCMA as the board supports only PCM. SPEEX is
disabled on the board.
-
Disabled the echo canceller. Did not have any effect on the result. -
In-call volume to increased 5.0x, 10.0x and went up to 100.0x.
Every time I increased this the audio gets worser.
-
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@01CDD3CC.69D07270
8-2-269/S/41 | Plot No.: 41 | Sagar Society| Banjara Hills | Road No.: 2 |
Hyderabad - 500038
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@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of Varma
Sent: Monday, April 07, 2014 9:04 AM
To: pjsip@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.
-
Tried using PCMU and PCMA as the board supports only PCM. SPEEX is
disabled on the board.
-
Disabled the echo canceller. Did not have any effect on the result. -
In-call volume to increased 5.0x, 10.0x and went up to 100.0x.
Every time I increased this the audio gets worser.
-
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@01CDD3CC.69D07270
8-2-269/S/41 | Plot No.: 41 | Sagar Society| Banjara Hills | Road No.: 2 |
Hyderabad - 500038
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@lists.pjsip.org
[mailto:pjsip-bounces@lists.pjsip.org] *On Behalf Of *Varma
Sent: Monday, April 07, 2014 9:04 AM
To: pjsip@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@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@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
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@lists.pjsip.org] On Behalf Of Bill Gardner
Sent: Thursday, April 10, 2014 7:35 PM
To: pjsip@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@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of Varma
Sent: Monday, April 07, 2014 9:04 AM
To: pjsip@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.
-
Tried using PCMU and PCMA as the board supports only PCM. SPEEX is
disabled on the board.
-
Disabled the echo canceller. Did not have any effect on the result. -
In-call volume to increased 5.0x, 10.0x and went up to 100.0x.
Every time I increased this the audio gets worser.
-
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@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@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
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@lists.pjsip.org] *On Behalf Of
*Bill Gardner
Sent: Thursday, April 10, 2014 7:35 PM
To: pjsip@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@lists.pjsip.org
<mailto:pjsip-bounces@lists.pjsip.org>
[mailto:pjsip-bounces@lists.pjsip.org] *On Behalf Of *Varma
*Sent:* Monday, April 07, 2014 9:04 AM
*To:* pjsip@lists.pjsip.org <mailto:pjsip@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@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/>
_______________________________________________
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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@lists.pjsip.org] On Behalf Of Bill Gardner
Sent: Tuesday, April 22, 2014 12:06 AM
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] 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@lists.pjsip.org] On Behalf Of Bill Gardner
Sent: Thursday, April 10, 2014 7:35 PM
To: pjsip@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@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of Varma
Sent: Monday, April 07, 2014 9:04 AM
To: pjsip@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.
-
Tried using PCMU and PCMA as the board supports only PCM. SPEEX is
disabled on the board.
-
Disabled the echo canceller. Did not have any effect on the result. -
In-call volume to increased 5.0x, 10.0x and went up to 100.0x.
Every time I increased this the audio gets worser.
-
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@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@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
