[sip-comm] Sip Communicator crashes at startup


#1

Laszlo,

It crashes at very early stage of VM startup (or we have a corrupted stack).

Check LD_LIBRARY_PATH and LD_PRELOAD vars in both environments (live
cd's and normal OS startup). Find a difference.

Take a look at linked libraries in worked environment (pmap the
process), find a difference with list from hs_err log. Make sure the
libraries in broken environment are exactly the same as they are in
worked.

regards,
dmeetry

Kertesz Laszlo wrote:

···

I have Ubuntu 9.04 installed on my computers (home and work computer). i
ran sip communicator development versions before, no problems. But for
some time i cannot launch sip communicator anymore. This is the terminal
output (the pagination seems a bit off, but thats the mai clients
fault):

#
# A fatal error has been detected by the Java Runtime Environment:
#
# SIGILL (0x4) at pc=0x8fc36922, pid=30658, tid=2418510736
#
# JRE version: 6.0_16-b01
# Java VM: Java HotSpot(TM) Server VM (14.2-b01 mixed mode linux-x86 )
# Problematic frame:
# C 0x8fc36922
#
# An error report file with more information is saved as:
# /home/laca/sip-communicator/hs_err_pid30658.log
#
# If you would like to submit a bug report, please visit:
# http://java.sun.com/webapps/bugreport/crash.jsp
#
./run.sh: line 4: 30658 Aborted java -classpath
"lib/jdic-all.jar:lib/jdic_stub.jar:lib/felix.jar:lib/bcprovider.jar:sc-bundles/sc-launcher.jar:sc-bundles/util.jar" -Djava.library.path=native -Dfelix.config.properties=file:./lib/felix.client.run.properties -Djava.util.logging.config.file=lib/logging.properties net.java.sip.communicator.launcher.SIPCommunicator

It happens on both of my desktop computers. I suspect it has to do with
some system settings (path or some applications) because sip-comm runs
if i start the computer from a live cd and it also runs on my laptop
with ubuntu 9.04.
I honestly dont know whats the difference between the computers configs,
all of them have the latest java installed from the repos and also other
java apps run just fine. I removed the .sip-communicator from my home
dir, uninstalled/reinstalled sip-communicator and it gives the same
(more or less) error every time.

Any thoughts on the cause this?

P.S. Attached the error log.

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#2

Ok, i finally sorted it out. It was a sound initialisation issue.

I use ALSA only (on both of the computers), no pulseaudio (i have
pulseaudio installed, but disabled on startup). But in the alsa
configuration file (/usr/share/alsa/alsa.conf) there is a line that
loads a pulseaudio related conf file (/usr/share/alsa/pulse.conf):

···

#
# ALSA library configuration file
#

# pre-load the configuration files

@hooks [
        {
                func load
                files [
                       "/usr/share/alsa/pulse.conf"

So, i commented out that last line, restarted alsa and that solved my
problem. Sip Communicator works now. Thanks for the reply.

Regards,
Kertesz Laszlo
              
On Wed, 2009-09-23 at 21:21 +0400, Dmeetry Degrave wrote:

Laszlo,

It crashes at very early stage of VM startup (or we have a corrupted stack).

Check LD_LIBRARY_PATH and LD_PRELOAD vars in both environments (live
cd's and normal OS startup). Find a difference.

Take a look at linked libraries in worked environment (pmap the
process), find a difference with list from hs_err log. Make sure the
libraries in broken environment are exactly the same as they are in
worked.

regards,
dmeetry

Kertesz Laszlo wrote:
> I have Ubuntu 9.04 installed on my computers (home and work computer). i
> ran sip communicator development versions before, no problems. But for
> some time i cannot launch sip communicator anymore. This is the terminal
> output (the pagination seems a bit off, but thats the mai clients
> fault):
>
> #
> # A fatal error has been detected by the Java Runtime Environment:
> #
> # SIGILL (0x4) at pc=0x8fc36922, pid=30658, tid=2418510736
> #
> # JRE version: 6.0_16-b01
> # Java VM: Java HotSpot(TM) Server VM (14.2-b01 mixed mode linux-x86 )
> # Problematic frame:
> # C 0x8fc36922
> #
> # An error report file with more information is saved as:
> # /home/laca/sip-communicator/hs_err_pid30658.log
> #
> # If you would like to submit a bug report, please visit:
> # http://java.sun.com/webapps/bugreport/crash.jsp
> #
> ./run.sh: line 4: 30658 Aborted java -classpath
> "lib/jdic-all.jar:lib/jdic_stub.jar:lib/felix.jar:lib/bcprovider.jar:sc-bundles/sc-launcher.jar:sc-bundles/util.jar" -Djava.library.path=native -Dfelix.config.properties=file:./lib/felix.client.run.properties -Djava.util.logging.config.file=lib/logging.properties net.java.sip.communicator.launcher.SIPCommunicator
>
>
> It happens on both of my desktop computers. I suspect it has to do with
> some system settings (path or some applications) because sip-comm runs
> if i start the computer from a live cd and it also runs on my laptop
> with ubuntu 9.04.
> I honestly dont know whats the difference between the computers configs,
> all of them have the latest java installed from the repos and also other
> java apps run just fine. I removed the .sip-communicator from my home
> dir, uninstalled/reinstalled sip-communicator and it gives the same
> (more or less) error every time.
>
> Any thoughts on the cause this?
>
> P.S. Attached the error log.

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#3

Hi,

I tried the .so that you sent me (i replaced the
default/usr/lib/sip-communicator/lib/native/libjportaudio.so). Build 2169. With the default .so the
mic wasnt working.
This test was done on Dell Optiplex 755 (Intel Corporation 82801I (ICH9 Family) HD Audio Controller)
/ Ubuntu 9.04 / ALSA only.

The recorded message from the server plays back perfectly, but once i hear back my input the sound
crackles/skips. Exactly as in the older versions.

Attached the log:
I ran sip-communicator --debug and called echo test 3 times - first all devices default, then i
changed the output devices to hw0:0. I couldnt hear anything when i had the out devices at hw 0:0,
but i still had network traffic (10 kb up/down).

Also the second hw device dissapeared. I had before 2 HDA Intel.... devices (i think meaning
front-out - on the front panel where i have my headset and back-out - the back panel's mic/out
connectors) now i have only 1 of them. I cannot see all of the device's names because they are
longer than the GUI allows to be seen - this is an issue too. I seen this behaviour (not showing the
second device) in some of the newer builds too.

P.S. Yes, i am from Romania. Are you from Russia?

Damian Minkov wrote:

···

Hi again,

as I said to you I've reproduced the problem, I've also took a look at
the changes and saw one thing that may impact this behavior. Can you
test the binary I'm sending to you and say does your capture device work
with it.
Also we are currently making big changes to media service. I'm also
continuing my work on portaudio and fixing several problems, one of
which is the one you reported. In the following weeks we will have
working noise reduction and echo cancellation. I've finally managed to
run them and successfully test them in sip-communicator, but there is
still many things to do before I manage to commit them and let people
test them, but and this will happen in next two weeks.
So after you test the binary I send please lets move our conversation to
the public :slight_smile: to the mailing list. I will personally write you there
when there is something you can test. (Emil, the project lead told me
that we must put our conversation to public and he is right, so others
can also see that we spend time doing this and that others can test it
too).

Thank you for the spent time and all the tests
Cheers
damencho

P.S. Are you situated in Romania?

Damian Minkov wrote:

yes, I know the commits are concerning other things, I managed to
reproduce your problem(just put my devices to be alsa ones not those
going through pulseaudio), but currently has other tasks to
complete(more urgent) and will try to fix that problem.

Thanks
again

Kertesz Laszlo wrote:

I have seen some new commits on the mailing list so i tried build
2158. The problem with the capture
is not yet solved (at least for me).

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#4

The sound doesnt work (mic and speakers) with ALSA on Ubuntu 9.04 if
other app is using audio (e.g. i browse the net and i watch videos in
flash, stop the video then call the echo test of our sip server, the
sound (at least output) fails to initialize ).
Terminal output:

11:35:10.805 SEVERE: impl.media.CallSessionImpl.controllerUpdate().2855
The following error was reported while starting a
playerjavax.media.ResourceUnavailableEvent[source=com.sun.media.content.unknown.Handler@1e54f45,message=Failed to prefetch: cannot open the audio device.]

Also if i listen music, there is no sound in sip communicator. Usually
when i stop the music (without quitting the application) i can use the
sound devices in sip communicator. But the flashplayer issue is really
unnerving cause i have to restart the browser every time i played
something with sound in it (and stopped the player, closed the tab). Not
to mention people who have the same problem.

Is it possible to make the java sound capture to be able to share the
output/input devices on ALSA (not sure how it works with pulseaudio, but
i suspect it is the same bhaviour)?


#5

Hi,

it seems the issue stays and the so file I sent doesn't fix it. It still seems like something blocks the input. But strange is how is it working in build 2140. Anyway, I will be reorganizing things and will change a little the way we use the devices/streams which I think will fix the problem. I will write you when there is something for testing. The problem is that from the log file I can only see what devices are used and nothing for the media flow. I will think of some mechanism to turn some debugging option in the native part so we can see is it actually reading/writing to devices so we can figure it out.

Thanks for the reports
damencho

Kertesz Laszlo wrote:

···

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi,

I tried the .so that you sent me (i replaced the
default/usr/lib/sip-communicator/lib/native/libjportaudio.so). Build 2169. With the default .so the
mic wasnt working.
This test was done on Dell Optiplex 755 (Intel Corporation 82801I (ICH9 Family) HD Audio Controller)
/ Ubuntu 9.04 / ALSA only.

The recorded message from the server plays back perfectly, but once i hear back my input the sound
crackles/skips. Exactly as in the older versions.

Attached the log:
I ran sip-communicator --debug and called echo test 3 times - first all devices default, then i
changed the output devices to hw0:0. I couldnt hear anything when i had the out devices at hw 0:0,
but i still had network traffic (10 kb up/down).

Also the second hw device dissapeared. I had before 2 HDA Intel.... devices (i think meaning
front-out - on the front panel where i have my headset and back-out - the back panel's mic/out
connectors) now i have only 1 of them. I cannot see all of the device's names because they are
longer than the GUI allows to be seen - this is an issue too. I seen this behaviour (not showing the
second device) in some of the newer builds too.

P.S. Yes, i am from Romania. Are you from Russia?

Damian Minkov wrote:
  

Hi again,

as I said to you I've reproduced the problem, I've also took a look at
the changes and saw one thing that may impact this behavior. Can you
test the binary I'm sending to you and say does your capture device work
with it.
Also we are currently making big changes to media service. I'm also
continuing my work on portaudio and fixing several problems, one of
which is the one you reported. In the following weeks we will have
working noise reduction and echo cancellation. I've finally managed to
run them and successfully test them in sip-communicator, but there is
still many things to do before I manage to commit them and let people
test them, but and this will happen in next two weeks.
So after you test the binary I send please lets move our conversation to
the public :slight_smile: to the mailing list. I will personally write you there
when there is something you can test. (Emil, the project lead told me
that we must put our conversation to public and he is right, so others
can also see that we spend time doing this and that others can test it
too).

Thank you for the spent time and all the tests
Cheers
damencho

P.S. Are you situated in Romania?

Damian Minkov wrote:
    

yes, I know the commits are concerning other things, I managed to
reproduce your problem(just put my devices to be alsa ones not those
going through pulseaudio), but currently has other tasks to
complete(more urgent) and will try to fix that problem.

Thanks
again

Kertesz Laszlo wrote:
      

I have seen some new commits on the mailing list so i tried build
2158. The problem with the capture
is not yet solved (at least for me).

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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

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#6

Hi,

do you have alsa-oss installed, cause with it such a problem can be avoided ?
The main problem is that java is using OSS for the sound input and output and if alsa-oss is installed it uses alsa.
I'm currently implementing a change where we will use portaudio (cross platform audio I/O library) which on linux will use alsa. With it we will have the option to choose more detailed the devices we want to use and will give us less latency on the audio output and capture.

damencho

Kertesz Laszlo wrote:

···

The sound doesnt work (mic and speakers) with ALSA on Ubuntu 9.04 if
other app is using audio (e.g. i browse the net and i watch videos in
flash, stop the video then call the echo test of our sip server, the
sound (at least output) fails to initialize ).
Terminal output:

11:35:10.805 SEVERE: impl.media.CallSessionImpl.controllerUpdate().2855
The following error was reported while starting a
playerjavax.media.ResourceUnavailableEvent[source=com.sun.media.content.unknown.Handler@1e54f45,message=Failed to prefetch: cannot open the audio device.]

Also if i listen music, there is no sound in sip communicator. Usually
when i stop the music (without quitting the application) i can use the
sound devices in sip communicator. But the flashplayer issue is really
unnerving cause i have to restart the browser every time i played
something with sound in it (and stopped the player, closed the tab). Not
to mention people who have the same problem.

Is it possible to make the java sound capture to be able to share the
output/input devices on ALSA (not sure how it works with pulseaudio, but
i suspect it is the same bhaviour)?

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#7

Hi,

I meant that the recording IS working now.
Only that the sound quality is poor like it was in the
older builds (up to 2140).

Sorry, it seems what i sent before was a bit confusing. My bad.

Damian Minkov wrote:

···

Hi,

it seems the issue stays and the so file I sent doesn't fix it. It still
seems like something blocks the input. But strange is how is it working
in build 2140. Anyway, I will be reorganizing things and will change a
little the way we use the devices/streams which I think will fix the
problem. I will write you when there is something for testing. The
problem is that from the log file I can only see what devices are used
and nothing for the media flow. I will think of some mechanism to turn
some debugging option in the native part so we can see is it actually
reading/writing to devices so we can figure it out.

Thanks for the reports
damencho

Kertesz Laszlo wrote:
Hi,

I tried the .so that you sent me (i replaced the
default/usr/lib/sip-communicator/lib/native/libjportaudio.so). Build
2169. With the default .so the
mic wasnt working.
This test was done on Dell Optiplex 755 (Intel Corporation 82801I
(ICH9 Family) HD Audio Controller)
/ Ubuntu 9.04 / ALSA only.

The recorded message from the server plays back perfectly, but once i
hear back my input the sound
crackles/skips. Exactly as in the older versions.

Attached the log:
I ran sip-communicator --debug and called echo test 3 times - first
all devices default, then i
changed the output devices to hw0:0. I couldnt hear anything when i
had the out devices at hw 0:0,
but i still had network traffic (10 kb up/down).

Also the second hw device dissapeared. I had before 2 HDA Intel....
devices (i think meaning
front-out - on the front panel where i have my headset and back-out -
the back panel's mic/out
connectors) now i have only 1 of them. I cannot see all of the
device's names because they are
longer than the GUI allows to be seen - this is an issue too. I seen
this behaviour (not showing the
second device) in some of the newer builds too.

P.S. Yes, i am from Romania. Are you from Russia?

Damian Minkov wrote:

Hi again,

as I said to you I've reproduced the problem, I've also took a look at
the changes and saw one thing that may impact this behavior. Can you
test the binary I'm sending to you and say does your capture device work
with it.
Also we are currently making big changes to media service. I'm also
continuing my work on portaudio and fixing several problems, one of
which is the one you reported. In the following weeks we will have
working noise reduction and echo cancellation. I've finally managed to
run them and successfully test them in sip-communicator, but there is
still many things to do before I manage to commit them and let people
test them, but and this will happen in next two weeks.
So after you test the binary I send please lets move our conversation to
the public :slight_smile: to the mailing list. I will personally write you there
when there is something you can test. (Emil, the project lead told me
that we must put our conversation to public and he is right, so others
can also see that we spend time doing this and that others can test it
too).

Thank you for the spent time and all the tests
Cheers
damencho

P.S. Are you situated in Romania?

Damian Minkov wrote:

yes, I know the commits are concerning other things, I managed to
reproduce your problem(just put my devices to be alsa ones not those
going through pulseaudio), but currently has other tasks to
complete(more urgent) and will try to fix that problem.

Thanks
again

Kertesz Laszlo wrote:

I have seen some new commits on the mailing list so i tried build
2158. The problem with the capture
is not yet solved (at least for me).

---------------------------------------------------------------------
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#8

Yes, i have alsa-oss installed. And if i launch Sip Communicator and its
idle, i cannot use audio input for other applications (such as skype).
There is a solution for this but it requires selecting "no device" in
sip comm every time to allow other applications use sound - but every
time u have to use sip-comm s audio, must stop the other apps, select
the javasound audio device again (And the flash player issue is the
biggest problem since you cannot stop it, only if u restart the
browser). Its not a real solution.
I hope implementing portaudio will solve these problems.

Regards

Kertesz Laszlo

···

On Fri, 2009-09-25 at 15:58 +0300, Damian Minkov wrote:

Hi,

do you have alsa-oss installed, cause with it such a problem can be
avoided ?
The main problem is that java is using OSS for the sound input and
output and if alsa-oss is installed it uses alsa.
I'm currently implementing a change where we will use portaudio (cross
platform audio I/O library) which on linux will use alsa. With it we
will have the option to choose more detailed the devices we want to use
and will give us less latency on the audio output and capture.

damencho

Kertesz Laszlo wrote:
> The sound doesnt work (mic and speakers) with ALSA on Ubuntu 9.04 if
> other app is using audio (e.g. i browse the net and i watch videos in
> flash, stop the video then call the echo test of our sip server, the
> sound (at least output) fails to initialize ).
> Terminal output:
>
> 11:35:10.805 SEVERE: impl.media.CallSessionImpl.controllerUpdate().2855
> The following error was reported while starting a
> playerjavax.media.ResourceUnavailableEvent[source=com.sun.media.content.unknown.Handler@1e54f45,message=Failed to prefetch: cannot open the audio device.]
>
> Also if i listen music, there is no sound in sip communicator. Usually
> when i stop the music (without quitting the application) i can use the
> sound devices in sip communicator. But the flashplayer issue is really
> unnerving cause i have to restart the browser every time i played
> something with sound in it (and stopped the player, closed the tab). Not
> to mention people who have the same problem.
>
> Is it possible to make the java sound capture to be able to share the
> output/input devices on ALSA (not sure how it works with pulseaudio, but
> i suspect it is the same bhaviour)?
>
>

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#9

Ahaa, I see.
That is also strange cause the so file has the all resampling stuff that must correct the sound problem. I will think of it :slight_smile:

Thanks
damencho

Kertesz Laszlo wrote:

···

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi,

I meant that the recording IS working now.
Only that the sound quality is poor like it was in the
older builds (up to 2140).

Sorry, it seems what i sent before was a bit confusing. My bad.

Damian Minkov wrote:
  

Hi,

it seems the issue stays and the so file I sent doesn't fix it. It still
seems like something blocks the input. But strange is how is it working
in build 2140. Anyway, I will be reorganizing things and will change a
little the way we use the devices/streams which I think will fix the
problem. I will write you when there is something for testing. The
problem is that from the log file I can only see what devices are used
and nothing for the media flow. I will think of some mechanism to turn
some debugging option in the native part so we can see is it actually
reading/writing to devices so we can figure it out.

Thanks for the reports
damencho

Kertesz Laszlo wrote:
Hi,

I tried the .so that you sent me (i replaced the
default/usr/lib/sip-communicator/lib/native/libjportaudio.so). Build
2169. With the default .so the
mic wasnt working.
This test was done on Dell Optiplex 755 (Intel Corporation 82801I
(ICH9 Family) HD Audio Controller)
/ Ubuntu 9.04 / ALSA only.

The recorded message from the server plays back perfectly, but once i
hear back my input the sound
crackles/skips. Exactly as in the older versions.

Attached the log:
I ran sip-communicator --debug and called echo test 3 times - first
all devices default, then i
changed the output devices to hw0:0. I couldnt hear anything when i
had the out devices at hw 0:0,
but i still had network traffic (10 kb up/down).

Also the second hw device dissapeared. I had before 2 HDA Intel....
devices (i think meaning
front-out - on the front panel where i have my headset and back-out -
the back panel's mic/out
connectors) now i have only 1 of them. I cannot see all of the
device's names because they are
longer than the GUI allows to be seen - this is an issue too. I seen
this behaviour (not showing the
second device) in some of the newer builds too.

P.S. Yes, i am from Romania. Are you from Russia?

Damian Minkov wrote:

Hi again,

as I said to you I've reproduced the problem, I've also took a look at
the changes and saw one thing that may impact this behavior. Can you
test the binary I'm sending to you and say does your capture device work
with it.
Also we are currently making big changes to media service. I'm also
continuing my work on portaudio and fixing several problems, one of
which is the one you reported. In the following weeks we will have
working noise reduction and echo cancellation. I've finally managed to
run them and successfully test them in sip-communicator, but there is
still many things to do before I manage to commit them and let people
test them, but and this will happen in next two weeks.
So after you test the binary I send please lets move our conversation to
the public :slight_smile: to the mailing list. I will personally write you there
when there is something you can test. (Emil, the project lead told me
that we must put our conversation to public and he is right, so others
can also see that we spend time doing this and that others can test it
too).

Thank you for the spent time and all the tests
Cheers
damencho

P.S. Are you situated in Romania?

Damian Minkov wrote:

yes, I know the commits are concerning other things, I managed to
reproduce your problem(just put my devices to be alsa ones not those
going through pulseaudio), but currently has other tasks to
complete(more urgent) and will try to fix that problem.

Thanks
again

Kertesz Laszlo wrote:

I have seen some new commits on the mailing list so i tried build
2158. The problem with the capture
is not yet solved (at least for me).

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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

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#10

Hmmm. It seems not all versions of Sip Communicator (i mean generic
java, distribution specific) work the same way. I was using the generic
java version. I could not use my webcam with that one and had the
aforementioned sound issues.

I recently installed the Ubuntu package (the launcher is more elaborate
and it preloads the alsa-oss libraries) and my webcam is working with
this version (and only this) and can listen to other audio when using
Sip Communicator...

It gets better and better...

···

On Sat, 2009-09-26 at 13:27 +0300, Kertesz Laszlo wrote:

Yes, i have alsa-oss installed. And if i launch Sip Communicator and its
idle, i cannot use audio input for other applications (such as skype).
There is a solution for this but it requires selecting "no device" in
sip comm every time to allow other applications use sound - but every
time u have to use sip-comm s audio, must stop the other apps, select
the javasound audio device again (And the flash player issue is the
biggest problem since you cannot stop it, only if u restart the
browser). Its not a real solution.
I hope implementing portaudio will solve these problems.

Regards

Kertesz Laszlo

On Fri, 2009-09-25 at 15:58 +0300, Damian Minkov wrote:
> Hi,
>
> do you have alsa-oss installed, cause with it such a problem can be
> avoided ?
> The main problem is that java is using OSS for the sound input and
> output and if alsa-oss is installed it uses alsa.
> I'm currently implementing a change where we will use portaudio (cross
> platform audio I/O library) which on linux will use alsa. With it we
> will have the option to choose more detailed the devices we want to use
> and will give us less latency on the audio output and capture.
>
> damencho
>
> Kertesz Laszlo wrote:
> > The sound doesnt work (mic and speakers) with ALSA on Ubuntu 9.04 if
> > other app is using audio (e.g. i browse the net and i watch videos in
> > flash, stop the video then call the echo test of our sip server, the
> > sound (at least output) fails to initialize ).
> > Terminal output:
> >
> > 11:35:10.805 SEVERE: impl.media.CallSessionImpl.controllerUpdate().2855
> > The following error was reported while starting a
> > playerjavax.media.ResourceUnavailableEvent[source=com.sun.media.content.unknown.Handler@1e54f45,message=Failed to prefetch: cannot open the audio device.]
> >
> > Also if i listen music, there is no sound in sip communicator. Usually
> > when i stop the music (without quitting the application) i can use the
> > sound devices in sip communicator. But the flashplayer issue is really
> > unnerving cause i have to restart the browser every time i played
> > something with sound in it (and stopped the player, closed the tab). Not
> > to mention people who have the same problem.
> >
> > Is it possible to make the java sound capture to be able to share the
> > output/input devices on ALSA (not sure how it works with pulseaudio, but
> > i suspect it is the same bhaviour)?
> >
> >
>
>
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#11

Hey Kertesz,

Java does not provide a platform independent way for video capture so we
don't support it in the generic java version.

... as a matter of fact we don't really maintain the generic java
version that much and keep it only as a proof of concept showing what
parts of SC could run (and how) without the use of native code.

Cheers,
Emil

Kertesz Laszlo wrote:

···

Hmmm. It seems not all versions of Sip Communicator (i mean generic
java, distribution specific) work the same way. I was using the generic
java version. I could not use my webcam with that one and had the
aforementioned sound issues.

I recently installed the Ubuntu package (the launcher is more elaborate
and it preloads the alsa-oss libraries) and my webcam is working with
this version (and only this) and can listen to other audio when using
Sip Communicator...

It gets better and better...

On Sat, 2009-09-26 at 13:27 +0300, Kertesz Laszlo wrote:

Yes, i have alsa-oss installed. And if i launch Sip Communicator and its
idle, i cannot use audio input for other applications (such as skype).
There is a solution for this but it requires selecting "no device" in
sip comm every time to allow other applications use sound - but every
time u have to use sip-comm s audio, must stop the other apps, select
the javasound audio device again (And the flash player issue is the
biggest problem since you cannot stop it, only if u restart the
browser). Its not a real solution.
I hope implementing portaudio will solve these problems.

Regards

Kertesz Laszlo

On Fri, 2009-09-25 at 15:58 +0300, Damian Minkov wrote:

Hi,

do you have alsa-oss installed, cause with it such a problem can be
avoided ?
The main problem is that java is using OSS for the sound input and
output and if alsa-oss is installed it uses alsa.
I'm currently implementing a change where we will use portaudio (cross
platform audio I/O library) which on linux will use alsa. With it we
will have the option to choose more detailed the devices we want to use
and will give us less latency on the audio output and capture.

damencho

Kertesz Laszlo wrote:

The sound doesnt work (mic and speakers) with ALSA on Ubuntu 9.04 if
other app is using audio (e.g. i browse the net and i watch videos in
flash, stop the video then call the echo test of our sip server, the
sound (at least output) fails to initialize ).
Terminal output:

11:35:10.805 SEVERE: impl.media.CallSessionImpl.controllerUpdate().2855
The following error was reported while starting a
playerjavax.media.ResourceUnavailableEvent[source=com.sun.media.content.unknown.Handler@1e54f45,message=Failed to prefetch: cannot open the audio device.]

Also if i listen music, there is no sound in sip communicator. Usually
when i stop the music (without quitting the application) i can use the
sound devices in sip communicator. But the flashplayer issue is really
unnerving cause i have to restart the browser every time i played
something with sound in it (and stopped the player, closed the tab). Not
to mention people who have the same problem.

Is it possible to make the java sound capture to be able to share the
output/input devices on ALSA (not sure how it works with pulseaudio, but
i suspect it is the same bhaviour)?

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--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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#12

I receive an error message and cannot hear the other end when connecting
to my workplace's VPN and logging in on the SIP server we have there.
The server runs TrixBox (essentially a customized Asterisk) and we
connect to it with Lynksys hard phones, and various soft SIP clients
[SIP Communicator (Ubuntu Linux), X Lite (Windows), Express Talk
(Windows), Ekiga (Ubuntu Linux), Linphone (Ubuntu Linux)]. We set up
OpenVPN connections to the network to be able to work from other
locations.
  The problem: I connect from home to the VPN.
-I have 192.168.100.100 IP on my local home net.
-The VPN creates a Tap interface with 192.168.5.x, creates routes
-On the work net the TrixBox server has 192.168.3.x

I dont know exactly the firewall rules, but it seems the addresses are
NAT ed so i can reach anything on the .3 network no problems.

-I can connect to the server with SIP Communicator, i can do echo test,
it works fine, i hear back my voice.

But when i want to talk with a some1 that is:
- Logged on his home net,
- VPN through to the work net, and
- he is using a Motorola e71 phone connected to the same SIP server with
its built in sip software (his local IP is 192.168.0.x), but NAT ed to
192.168.3.x (he has a bit different VPN connection, it connects directly
to the .3 class net not the .5 as me).
results in:

-I cant hear him, but he hears me.

Other thing to consider is that i monitor my net traffic and have a
steady incoming/outgoing 12-13 KB traffic during the connection. This
means that i do receive the packets but the program fails to process
them (i dont know why).

On the other hand, on my both my work (directly connected to the
internal network) and my home computer connected through VPN i receive
the same error messages:

11:28:07.126 SEVERE:
impl.protocol.sip.SipStackSharing.processRequest().583 couldn't create
transaction, please report this to dev@sip-communicator.dev.java.net
javax.sip.TransactionUnavailableException: Missing a required header :
Subscription-State
  at
gov.nist.javax.sip.SipProviderImpl.getNewServerTransaction(SipProviderImpl.java:459)
  at
net.java.sip.communicator.impl.protocol.sip.SipStackSharing.processRequest(SipStackSharing.java:567)
  at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:223)
  at gov.nist.javax.sip.EventScanner.run(EventScanner.java:492)
  at java.lang.Thread.run(Thread.java:619)
Caused by: java.text.ParseException: Missing a required header :
Subscription-State
  at
gov.nist.javax.sip.message.SIPRequest.checkHeaders(SIPRequest.java:239)
  at
gov.nist.javax.sip.SipProviderImpl.getNewServerTransaction(SipProviderImpl.java:457)
  ... 4 more

And it goes on, about 1 message/sec or something like it.

What is the problem here? I do receive the data but i cant actually hear
it (echo test on the other hand works fine). Is it a software
incompatibility or SIP problem that has to do with VPN because if we
both are connected to the work network it works fine?


#13

I tested Sip Communicator (build 2097/2098) and i have come accross some
bugs
(or issues).

Environment:
OS: Ubuntu 9.04 (and 8.10), running ALSA only.

- - internal network with internet access (firewalled);
- - TrixBox SIP server that is connected to a PSTN phone line aswell;
- - OpenFire jabber server for internal messaging;
- - Other jabber clients on the network:
   - Pidgin (Windows/Linux versions, used by everyone) and
   - Empathy (this one is not actually used widely, but did tests with it)
- - VPN connections - all of us have VPN (OpenVPN) remote access to the
internal network resources;
- - Yahoo, Google Talk, Jabber, SIP protocols tested.

Working stuff:

- - Text message sending/receiving on all messaging protocols;
- - Yahoo file transfer (receiving/transmitting);
- - Google Talk file transfer (receiving/transmitting)
- - SIP calling - *when connected to the local network only* (also works
with calling via the PSTN line)
- - Webcam [i tested only with ekiga test call, it seems to work (i havent
tried an actual call), albeit with a bit of lag]

Issues (meaning that features that dont work as they should be)

- - Jabber file transfer:
Cannot send file from Pidgin/Empathy to Sip Communicator (they say Sip
Communicator doesnt have file transfer capabilities. It can, however, to
send file sometimes to Pidgin. All in all it really doesnt work.
I might add that Pidgin-Pidgin, Empathy-Pidgin file transfer works (at
really high speeds).

- - PortAudio (i even edited the launcher script and commented out the
alsa-oss preloading, but it had no effect on the result):
  - Selecting other device (ex. HW 0:0) than "default" or selecting java
sound when previously portaudio (with HW 0:0 input device) was selected
hangs the X session (have to kill java from tty1) but it doesnt lock X
completely, just steals focus and disables mouse clicking/shortcuts
(other than ctrl-alt-backspace). Restarting sip communicator after that
i have the device i wanted.
  - The "default" capture device doesnt work. HW 0:0 works.
  - Sound quality is low (high pitch). And the sound crackles sometimes.
Low sampling rate?

- SIP (mainly the issues are bound to VPN connections, on the local net
it works fine) :

  - If the Sip Communicator client is connected via VPN (there is a NAT
involved if that helps) it cannot hear the other end (although it is a
constant ~10KB up/down data transfer as in cases when everything works),
but the other can hear the Sip Communicator. But on such case that the
Sip Communicator client is on the local net and the other client is
connected through VPN it works [but tested only with a VPN client that
connects with the same network address (192.168.3.x) as the local net,
the most used VPNs have the .5 class NAT ed to .3 ].

  - Lots of console errors of the type:
  
12:23:59.025 SEVERE:
impl.protocol.sip.SipStackSharing.processRequest().583 couldn't create
transaction, please report this to dev@sip-communicator.dev.java.net
javax.sip.TransactionUnavailableException: Missing a required header :
Subscription-State
  at
gov.nist.javax.sip.SipProviderImpl.getNewServerTransaction(SipProviderImpl.java:459)
  at
net.java.sip.communicator.impl.protocol.sip.SipStackSharing.processRequest(SipStackSharing.java:567)
  at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:223)
  at gov.nist.javax.sip.EventScanner.run(EventScanner.java:492)
  at java.lang.Thread.run(Thread.java:619)
Caused by: java.text.ParseException: Missing a required header :
Subscription-State
  at gov.nist.javax.sip.message.SIPRequest.checkHeaders(SIPRequest.java:239)
  at
gov.nist.javax.sip.SipProviderImpl.getNewServerTransaction(SipProviderImpl.java:457)
  ... 4 more
12:23:59.025 SEVERE:
impl.protocol.sip.SipStackSharing.logApplicationException().968 An error
occurred while processing event of type: javax.sip.DialogTerminatedEvent

These are the issues i came accross for now.

···

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#14

Hi,

Kertesz Laszlo wrote:

-----BEGIN PGP SIGNED MESSAGE-----
Environment:
OS: Ubuntu 9.04 (and 8.10), running ALSA only.

- - PortAudio (i even edited the launcher script and commented out the
alsa-oss preloading, but it had no effect on the result):
  - Selecting other device (ex. HW 0:0) than "default" or selecting java
sound when previously portaudio (with HW 0:0 input device) was selected
hangs the X session (have to kill java from tty1) but it doesnt lock X
completely, just steals focus and disables mouse clicking/shortcuts
(other than ctrl-alt-backspace). Restarting sip communicator after that
i have the device i wanted.
  - The "default" capture device doesnt work. HW 0:0 works.
  - Sound quality is low (high pitch). And the sound crackles sometimes.
Low sampling rate?
  

About the problems you have, can you give me more info about your configuration,
cause I'm currently investigating such issues :
- kernel version
- alsa version
- portaudio version
- pulseaudio is it installed and if yes the version
- sound card model

About the default device I got one and I got it only when pulseaudio is installed (this is default on Ubuntu and gnome), but I see on their webpage http://www.pulseaudio.org/wiki/BrokenSoundDrivers that my driver is buggy (snd-intel-hda). I also get those X locks when switching from default. But when I uninstall pulseaudio I can freely switch among all devices.

Thanks
damencho

···

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#15

Damian Minkov wrote:

Hi,

Kertesz Laszlo wrote:

-----BEGIN PGP SIGNED MESSAGE-----
Environment:
OS: Ubuntu 9.04 (and 8.10), running ALSA only.

- - PortAudio (i even edited the launcher script and commented out the
alsa-oss preloading, but it had no effect on the result):
    - Selecting other device (ex. HW 0:0) than "default" or selecting
java
sound when previously portaudio (with HW 0:0 input device) was selected
hangs the X session (have to kill java from tty1) but it doesnt lock X
completely, just steals focus and disables mouse clicking/shortcuts
(other than ctrl-alt-backspace). Restarting sip communicator after that
i have the device i wanted.
    - The "default" capture device doesnt work. HW 0:0 works.
    - Sound quality is low (high pitch). And the sound crackles
sometimes.
Low sampling rate?
  

About the problems you have, can you give me more info about your
configuration,
cause I'm currently investigating such issues :
- kernel version
- alsa version
- portaudio version
- pulseaudio is it installed and if yes the version
- sound card model

About the default device I got one and I got it only when pulseaudio is
installed (this is default on Ubuntu and gnome), but I see on their
webpage http://www.pulseaudio.org/wiki/BrokenSoundDrivers that my driver
is buggy (snd-intel-hda). I also get those X locks when switching from
default. But when I uninstall pulseaudio I can freely switch among all
devices.

Thanks
damencho

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Hi,

My specs
Hardware: Dell Optiplex 755 on 1 comp/ ASUS M3N78-VM mobo + AMD 3200+ proc on the other
Sound card:
Dell (lspci): 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev
02) (snd_hda_intel module is used),
Alsamixer: Card: HDA Intel, Chip: Analog Devices AD1984

ASUS: I am not at that computer at the moment, but i found on the net the following:
    00:07.0 Audio device: nVidia Corporation MCP78S [GeForce 8200] High Definition Audio (rev a1), i
suspect it uses the snd_hda_intel module aswell.

OS: Ubuntu 9.04
Kernel: 2.6.28-15-generic
PulseAudio: 1:0.9.14-0ubuntu20 (0.9.14 is the real version i guess)
ALSA: 1.0.18
PortAudio: I have libportaudio version 18.1-7.1 installed. Is this what Sip Communicator is using?

I experience the same low quality/choppy sound on both computers (JavaSound is better quality, but
doesnt compare to native apps such as linphone and skype using ALSA and it locks the input device
aswell). And cannot use the default device as input only the HW 0:0. This means other apps cannot
use the mic device while Sip Communicator is running.

PulseAudio is installed (Version: 1:0.9.14-0ubuntu20 , default ubuntu 9.04, no custom builds, buggy
as hell) but i unload it (i have a killall pulseaudio startup script (starts on logon)) and i set
the pulseaudio script (client.conf) to respawn=no so that it will not reload.

Another thing i came across (not sound related): i receive Yahoo Messenger messages duplicated (at 7
seconds interval) if in the meantime i dont send something. This happens only if the other end uses
Yahoo Messenger on windows as far as i experienced it.

Regards,

Kertesz Laszlo

···

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#16

Damian Minkov wrote:

Hi,

My specs
Hardware: Dell Optiplex 755 on 1 comp/ ASUS M3N78-VM mobo + AMD 3200+ proc on the other
Sound card:
Dell (lspci): 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev
02) (snd_hda_intel module is used),
Alsamixer: Card: HDA Intel, Chip: Analog Devices AD1984

ASUS: I am not at that computer at the moment, but i found on the net the following:
    00:07.0 Audio device: nVidia Corporation MCP78S [GeForce 8200] High Definition Audio (rev a1), i
suspect it uses the snd_hda_intel module aswell.

OS: Ubuntu 9.04
Kernel: 2.6.28-15-generic
PulseAudio: 1:0.9.14-0ubuntu20 (0.9.14 is the real version i guess)
ALSA: 1.0.18
PortAudio: I have libportaudio version 18.1-7.1 installed. Is this what Sip Communicator is using?

I experience the same low quality/choppy sound on both computers (JavaSound is better quality, but
doesnt compare to native apps such as linphone and skype using ALSA and it locks the input device
aswell). And cannot use the default device as input only the HW 0:0. This means other apps cannot
use the mic device while Sip Communicator is running.

PulseAudio is installed (Version: 1:0.9.14-0ubuntu20 , default ubuntu 9.04, no custom builds, buggy
as hell) but i unload it (i have a killall pulseaudio startup script (starts on logon)) and i set
the pulseaudio script (client.conf) to respawn=no so that it will not reload.

Another thing i came across (not sound related): i receive Yahoo Messenger messages duplicated (at 7
seconds interval) if in the meantime i dont send something. This happens only if the other end uses
Yahoo Messenger on windows as far as i experienced it.

Regards,

Kertesz Laszlo

···

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#17

Hi,

In the recent alphas portaudio shows only /dev/dsp device on all 3 sections. I think it started
since the layout was slightly changed. Before i have seen the ALSA specific devices (HW 0.0,
default, etc). Now i have only /dev/dsp. And it doesnt work at all. No sound. JavaSound is working.

Kertesz Laszlo

···

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#18

I can't get the SIP communicator source code as a netbeans project.I
followed the documentation by Brian Burch and still can't get it to work.

Can anyone help me?

Thank,s
Komin


#19

Hi,

thank you for all you reports, event I'm not responding to all of them they are all very useful :slight_smile:
I know about this problem. I integrated a patch into portaudio which aim was to avoid segmentation faults on some systems when pulse audio is installed and working. But it seems to filter all alsa devices, strange to me is that it happens on 32bit systems and not on my amd64 ubuntu install. I hope I will fix this these days.

Thanks
damencho

Kertesz Laszlo wrote:

···

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi,

In the recent alphas portaudio shows only /dev/dsp device on all 3 sections. I think it started
since the layout was slightly changed. Before i have seen the ALSA specific devices (HW 0.0,
default, etc). Now i have only /dev/dsp. And it doesnt work at all. No sound. JavaSound is working.

Kertesz Laszlo
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.9 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

iEYEARECAAYFAkrcgSsACgkQrlZ/rySPZHru+gCeLH28SFm2aZ0ukA655RoZuVYb
FQcAnj5O4lLM10zJ+cXJC9euwYfdTQBj
=DQGb
-----END PGP SIGNATURE-----

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#20

Damian Minkov wrote:

Hi,

thank you for all you reports, event I'm not responding to all of them
they are all very useful :slight_smile:
I know about this problem. I integrated a patch into portaudio which aim
was to avoid segmentation faults on some systems when pulse audio is
installed and working. But it seems to filter all alsa devices, strange
to me is that it happens on 32bit systems and not on my amd64 ubuntu
install. I hope I will fix this these days.

Thanks
damencho

Ok, i got it. I am curious about this PortAudio thing because i have seen it in other programs
(sip/iax softphones like kiax and voixphone), but it didnt quite work right except one occasion
(voixphone), but then i had to preload ala-oss to do so - it has the same problem as the one
integrated in Sip Communicator, it sees only /dev/dsp, but i have good sound quality there.

I have identified a few major bugs (or missing features perhaps?) in Sip Communicator that (at least
for me) makes impossible to use it for everything in this stage (i have2 Yahoo accounts, 1 local
Jabber account, 1 gmail account, 1 local SIP account that i use). And i have an IAX account too.
By the way, is there a possibility to implement IAX/IAX2 support in Sip Communicator?

The big problems:

- - Jabber file transfer on the local (intranet) server (OpenFire) - some other other clients (Pidgin
and Empathy) wont send files to SIP Communicator reporting "No file transfer capabilities". Recently
i have tested Psi and has no problems in sending/receiving files to/from Sip Communicator.
  But being the case that the others around here use Pidgin, i cannot disregard this fact. I must log
in from another client with higher priority when i am to receive a file. I think it is related to
different/non standard XMPP/Jabber implementation in different programs, but is it possible to have
a look at the Jabber implementation in Sip communicator to address this issue? Because many people
use Pidgin.

- - I cannot make a call whem im logging in from home via VPN and connecting to our SIP Server. I have
mentioned this issue before. The other end can hear me, but i cant hear him.
  I am not sure that it has to do with SIP/NAT traversal issues, because i have incoming/outgoing
stream (constant ~15 KB/s up/down). I can succesfully make test calls however (Its a
TrixBox/Asterisk server). Other clients (Zoiper 2.13 and Linphone 3.2.1) however work in this scenario.

Smaller/annoying problems:

- - Yahoo messenger duplicate messages when the other end uses THE Yahoo Messenger (i read something
that it has to do with lack of confirmation of receiving the message in 7 seconds - it doesnt happen
if i send something right after i receive the message).
- - Sip Communicator uses 100% CPU (and i think the sound is broken in meantime) if on the
Options->Notifications tab i click on sound previews at less than about 2 seconds interval. I have
to restart the application.

···

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