Hello All
Thanks for the prompt replies.
I am using pjsip 0.9.0 version on openmoko FR compiled in arm-angstrom-linux environment
by using following command
--host=arm-angstrom-linux-gnueabi
I am having problem with sound quality
* audio stutters,
* audio break-ups. I looked through the checklist https://trac.pjsip.org/repos/wiki/sound-problems, and I ran the sound test by
./sndtest (https://trac.pjsip.org/repos/wiki/audio-check-sound-device-jitter)
and got the below given results .
As the avg jitter is below 32 ms so that is fine but the max jitter is very high (132.168ms) so as suggested on that page
I would like to ask what can be the problem. Do I need to change the jitter buffer settings according to device (openmoko FR).?
Further I tried using PCMU and Speex codecs but do not see any change in CPU load.
Any kind of suggestion will be greatly appeciated.
Thanks
Tarun
----- Original Message ----
From: Benny Prijono bennylp@pjsip.org
To: pjsip list pjsip@lists.pjsip.org
Sent: Monday, August 18, 2008 8:29:01 PM
Subject: Re: [pjsip] how to reduce CPU load and improve voice quality on mobile device FR
On Mon, Aug 18, 2008 at 6:51 AM, TARUN CHAPLOT tarunchaplot@yahoo.com wrote:
Hi,
We are having a problem with running PJSIP on the neo freerunner, and have tried a
variety of solutions to fix this but still not successful. The setup
that we are using is PJSIP on a Mac with OSX, and PJSIP compiled for
openmoko.I compiled PJSIP for openmoko by setting the env using command
source /usr/local/openmoko/arm/setup-env
and ./configure --host=arm-angstrom-linux-gnueabi.
First thing first, what pjsip version?
When we run PJSIP between the Mac and Openmoko, we are encountering problems.
The problem is that the sound quality is very bad. When we speak
into the Mac, the sound coming from the Openmoko comes out in stutters
(on and off every half second). It is also full of static.
When we try a VOIP call directly from mac to mac, it works and sounds fine.
If you haven't seen this, perhaps you can follow the checklists in https://trac.pjsip.org/repos/wiki/sound-problems, and find out what the exact problem is.
I looked at the CPU load and found that it is almost 80% on freerunner and 25 % on Mac OS X.
I have tried with different codecs e.g. speex ,PCMA and GSM.
Please advise me that how can I reduce the CPU load. I have followed the following http://trac.pjsip.org/repos/wiki/FAQ#cpu but
no improvement.
Now that's surprising. You should see significant difference in CPU usage between Speex and PCMU. If not, then that probably suggests that the problem is elsewhere.
Cheers
Benny
Thanks in advance.
Tarun
On Mon, Aug 18, 2008 at 6:14 PM, TARUN CHAPLOT tarunchaplot@yahoo.comwrote:
Hello All
Thanks for the prompt replies.
I am using pjsip 0.9.0 version on openmoko FR compiled in
arm-angstrom-linux environment
by using following command
--host=arm-angstrom-linux-gnueabi
I've no clue about OpenMoko so I'm not sure what build command to suggest,
but I guess if it builds and runs fine then it's okay then. More below.
I am having problem with sound quality
- audio stutters,
- audio break-ups.
I looked through the checklist
https://trac.pjsip.org/repos/wiki/sound-problems, and I ran the sound test
by
./sndtest (https://trac.pjsip.org/repos/wiki/audio-check-sound-device-jitter)
and got the below given results .
17:49:15.121 os_core_unix.c pjlib 0.9.0-release for POSIX initialized
17:49:19.212 pasound.c PortAudio sound library initialized, status=0
17:49:19.215 pasound.c PortAudio host api count=1
17:49:19.217 pasound.c Sound device count=1
17:49:19.222 pjlib select() I/O Queue created (0x47754)
17:49:19.226 sndtest.c Found 1 devices:
17:49:19.228 sndtest.c 0: /dev/dsp (capture=16, playback=16)
17:49:19.915 sndtest.c Testing playback device /dev/dsp
17:49:19.917 sndtest.c Testing capture device /dev/dsp
17:49:20.481 sndtest.c Please wait while test is in progress (~11
secs)..
17:49:32.424 sndtest.c Dumping results:
17:49:32.425 sndtest.c Parameters: clock rate=8000Hz, 80
samples/frame
17:49:32.425 sndtest.c Playback stream report:
17:49:32.425 sndtest.c Duration: 9s.980
17:49:32.426 sndtest.c Frame interval: min=0.004ms, max=132.173ms
17:49:32.426 sndtest.c Jitter: min=9.955ms, avg=28.194ms,
max=132.168ms
17:49:32.427 sndtest.c Capture stream report:
17:49:32.427 sndtest.c Duration: 9s.980
17:49:32.427 sndtest.c Frame interval: min=0.003ms, max=132.144ms
17:49:32.428 sndtest.c Jitter: min=9.943ms, avg=28.187ms,
max=132.141ms
17:49:32.428 sndtest.c Checking for clock drifts:
17:49:32.428 sndtest.c No clock drifts is detected
17:49:32.429 sndtest.c Test completed with some warnings
As the avg jitter is below 32 ms so that is fine but the max jitter is very
high (132.168ms) so as suggested on that page
That's awful indeed. You can try the following (either alone or combination
of them) and see which one helps:
I would like to ask what can be the problem. Do I need to change the jitter
buffer settings according to device (openmoko FR).?
The problem is with the sound device itself. I'm not sure changing jitter
buffer settings will help since the default values should be big enough
already (360ms). But lets do the above first and we'll come back at this
later.
Further I tried using PCMU and Speex codecs but do not see any change in
CPU load.
The FAQ http://trac.pjsip.org/repos/wiki/FAQ#cpu really has explained
everything, if you don't see any improvements I suspect some of the steps
are not followed properly. In particular please read again item no 4 there
about choosing sampling rate, which means that if you use PCMU, you should
also use --clock-rate 8000 option.
Cheers
Benny
Any kind of suggestion will be greatly appeciated.
Thanks
Tarun
----- Original Message ----
From: Benny Prijono bennylp@pjsip.org
To: pjsip list pjsip@lists.pjsip.org
Sent: Monday, August 18, 2008 8:29:01 PM
Subject: Re: [pjsip] how to reduce CPU load and improve voice quality on
mobile device FR
On Mon, Aug 18, 2008 at 6:51 AM, TARUN CHAPLOT tarunchaplot@yahoo.comwrote:
Hi,
We are having a problem with running PJSIP on the neo freerunner, and
have tried a
variety of solutions to fix this but still not successful. The setup
that we are using is PJSIP on a Mac with OSX, and PJSIP compiled for
openmoko.I compiled PJSIP for openmoko by setting the env using command
source /usr/local/openmoko/arm/setup-env
and ./configure --host=arm-angstrom-linux-gnueabi.
First thing first, what pjsip version?
When we run PJSIP between the Mac and Openmoko, we are encountering
problems.
The problem is that the sound quality is very bad. When we speak
into the Mac, the sound coming from the Openmoko comes out in stutters
(on and off every half second). It is also full of static.
When we try a VOIP call directly from mac to mac, it works and sounds
fine.
If you haven't seen this, perhaps you can follow the checklists in
https://trac.pjsip.org/repos/wiki/sound-problems, and find out what the
exact problem is.
I looked at the CPU load and found that it is almost 80% on freerunner and
25 % on Mac OS X.
I have tried with different codecs e.g. speex ,PCMA and GSM.
Please advise me that how can I reduce the CPU load. I have followed the
following http://trac.pjsip.org/repos/wiki/FAQ#cpu but
no improvement.
Now that's surprising. You should see significant difference in CPU usage
between Speex and PCMU. If not, then that probably suggests that the problem
is elsewhere.
Cheers
Benny
Thanks in advance.
Tarun
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