how to configure app_conference??
i follow the instruction on voip-info.org but the process still
produce a
lot of error.
You might want to start by reading the README file, followed by
CLI.txt and Dialplan.txt. These files are packaged with appconference
source.
Make sure you are using a supported codec. Make sure you're using
Asterisk 1.4.x.
Appconference is not hard to set up if you take a few minutes and do
your research.
If you are still not able to get it to work, you can ping the
appconference developer list on sourceforge.net
Mihai
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Posted: Wed May 14, 2008 8:20 pm Post subject: [Asterisk-video] how to configure app_conference
The first time I try to install the app_conference produce a lot of error too, but with some change in the paths of the Makefile, I install normaly.
----- Mensaje original ----
De: Mihai Balea <mihai@hates.ms>
Para: Development discussion of video media support in Asterisk <asterisk-video@lists.digium.com>
Enviado: lunes, 12 de mayo, 2008 16:13:12
Asunto: Re: [Asterisk-video] how to configure app_conference
how to configure app_conference??
i follow the instruction on voip-info.org but the process still
produce a
lot of error.
You might want to start by reading the README file, followed by
CLI.txt and Dialplan.txt. These files are packaged with appconference
source.
Make sure you are using a supported codec. Make sure you're using
Asterisk 1.4.x.
Appconference is not hard to set up if you take a few minutes and do
your research.
If you are still not able to get it to work, you can ping the
appconference developer list on sourceforge.net
Mihai
Enviado desde Correo Yahoo!
La bandeja de entrada más inteligente.
Posted: Thu May 15, 2008 7:04 am Post subject: [Asterisk-video] how to configure app_conference
Hi,
I have a binary package (version 2.0.1 without modifications, deliver
the full original sources too).
I think, it can help you to install quickly the application if your have
problems when compiling the sources.
Download it from the www.i6net.com web site.
It deliver the app_conference.so compiled with Asterisk 1.4.19 in a
32bits platform.
Unzip/Untar, execute install.sh and restart your Asterisk.
The voice conference works well, but I didn't able to use the video
switching based on VAD or DTMF.
The first time I try to install the app_conference produce a lot of
error too, but with some change in the paths of the Makefile, I
install normaly.
----- Mensaje original ----
De: Mihai Balea <mihai@hates.ms>
Para: Development discussion of video media support in Asterisk
<asterisk-video@lists.digium.com>
Enviado: lunes, 12 de mayo, 2008 16:13:12
Asunto: Re: [Asterisk-video] how to configure app_conference
> hy all
>
> how to configure app_conference??
> i follow the instruction on voip-info.org but the process still
> produce a
> lot of error.
You might want to start by reading the README file, followed by
CLI.txt and Dialplan.txt. These files are packaged with appconference
source.
Make sure you are using a supported codec. Make sure you're using
Asterisk 1.4.x.
Appconference is not hard to set up if you take a few minutes and do
your research.
If you are still not able to get it to work, you can ping the
appconference developer list on sourceforge.net
Enviado desde Correo Yahoo!
<http://us.rd.yahoo.com/mailuk/taglines/isp/control/*http://us.rd.yahoo.com/evt=52431/*http://es.docs.yahoo.com/mail/overview/index.html>
La bandeja de entrada más inteligente.
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Posted: Thu May 15, 2008 9:36 am Post subject: [Asterisk-video] how to configure app_conference
On May 15, 2008, at 3:53 AM, Borja SIXTO wrote:
Quote:
The voice conference works well, but I didn't able to use the video
switching based on VAD or DTMF.
Just a quick note here, since this is one of the more confusing
aspects of appconference:
To get VAD-based video switching working, your clients need to use
DTX (i.e. stop sending audio packets when there is no voice
activity). While it is possible to implement a full VAD solution in
appconference (the relevant code is there), VAD is quite CPU-
intensive, so there are advantages to offloading this to the clients.
Mihai
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Posted: Thu May 15, 2008 9:56 am Post subject: [Asterisk-video] how to configure app_conference
Hi Mihai,
For the 3G handsets we cannot configure this mode for example (Some
phones generate always an audio stream).
So I think it will be great to be able to enable the full VAD with a
specific application option.
I will read again the source code.
Thanks,
Tech from i6net
Mihai Balea a écrit :
Quote:
On May 15, 2008, at 3:53 AM, Borja SIXTO wrote:
>
> The voice conference works well, but I didn't able to use the video
> switching based on VAD or DTMF.
Just a quick note here, since this is one of the more confusing
aspects of appconference:
To get VAD-based video switching working, your clients need to use
DTX (i.e. stop sending audio packets when there is no voice
activity). While it is possible to implement a full VAD solution in
appconference (the relevant code is there), VAD is quite
CPU-intensive, so there are advantages to offloading this to the clients.
Mihai
------------------------------------------------------------------------
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Posted: Thu May 15, 2008 10:51 am Post subject: [Asterisk-video] how to configure app_conference
Hi Borja,
Having full VAD in appconference is actually a feature I am interested
in -- I just didn't have the time to do it.
It should actually be quite simple to implement - if you want to go
ahead and attempt to do it, I would be more than happy to merge in
your patch.
Here are a few pointers that might help you understand what needs to
be done:
- the place you want to look at is member.[ch]
- if you look at create_member(), you will notice that members marked
as 't' (telephone) have a speex preprocessor instance associated with
them. As the code stands right now, members coming in via a zaptel
connection cannot do DTX (obviously), so appconference does VAD, AGC
and noise reduction by itself.
- however, if a member is marked as 't', it will not be considered as
a candidate for video switching (again, for obvious reasons - no video
on zaptel connections).
So what would probably need to be done is define another dialplan
flag, that would make appconference initialize a DSP engine for that
member. It would be similar to 't', but due to backwards compatibility
issues, we cannot really use 't' for this purpose. Once that is done,
appconference _should_ be aware of voice activity from members that do
not do DTX and _should_ be able to include this information in the VAD-
based video switching logic. It will obviously need to be tested and
there is a good chance that some other parts of the code will need to
be tweaked.
Hope this helps. Let me know if you have any questions. I have also
copied the appconference dev list on this, and I suggest that we move
this discussion there, since it is appconference-specific and doesn't
really relate to asterisk video in general.
Mihai
On May 15, 2008, at 6:48 AM, Borja SIXTO wrote:
Quote:
Hi Mihai,
For the 3G handsets we cannot configure this mode for example (Some
phones generate always an audio stream).
So I think it will be great to be able to enable the full VAD with a
specific application option.
I will read again the source code.
Thanks,
Tech from i6net
Mihai Balea a écrit :
>
> On May 15, 2008, at 3:53 AM, Borja SIXTO wrote:
>>
>> The voice conference works well, but I didn't able to use the video
>> switching based on VAD or DTMF.
>
> Just a quick note here, since this is one of the more confusing
> aspects of appconference:
> To get VAD-based video switching working, your clients need to use
> DTX (i.e. stop sending audio packets when there is no voice
> activity). While it is possible to implement a full VAD solution in
> appconference (the relevant code is there), VAD is quite
> CPU-intensive, so there are advantages to offloading this to the
> clients.
>
> Mihai
> ------------------------------------------------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
_______________________________________________
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Posted: Mon May 19, 2008 8:03 pm Post subject: [Asterisk-video] how to configure app_conference
Hi Hihai,
I have added the DSP and silence detector.
But now, I cannot make a nice video switching...
I would like to control cleanly the video switching.
Actually it is strange.
I need to know how do you select the video member
(conf->current_video_source_id).
The video source should be unselected if any one in the conference is
speaking and the source is in audio silence.
If not, the last speaker is selected waiting for some one who will start
to speech.
An other interesting function will be : use "#" to get the focus.
If I would like to speak, I press the "#" in order to get the focus and
start speaking.
What do you think about ?
Regards,
Tech from i6net
Mihai Balea a écrit :
Quote:
Hi Borja,
Having full VAD in appconference is actually a feature I am interested
in -- I just didn't have the time to do it.
It should actually be quite simple to implement - if you want to go
ahead and attempt to do it, I would be more than happy to merge in
your patch.
Here are a few pointers that might help you understand what needs to
be done:
- the place you want to look at is member.[ch]
- if you look at create_member(), you will notice that members marked
as 't' (telephone) have a speex preprocessor instance associated with
them. As the code stands right now, members coming in via a zaptel
connection cannot do DTX (obviously), so appconference does VAD, AGC
and noise reduction by itself.
- however, if a member is marked as 't', it will not be considered as
a candidate for video switching (again, for obvious reasons - no video
on zaptel connections).
So what would probably need to be done is define another dialplan
flag, that would make appconference initialize a DSP engine for that
member. It would be similar to 't', but due to backwards compatibility
issues, we cannot really use 't' for this purpose. Once that is done,
appconference _should_ be aware of voice activity from members that do
not do DTX and _should_ be able to include this information in the
VAD-based video switching logic. It will obviously need to be tested
and there is a good chance that some other parts of the code will need
to be tweaked.
Hope this helps. Let me know if you have any questions. I have also
copied the appconference dev list on this, and I suggest that we move
this discussion there, since it is appconference-specific and doesn't
really relate to asterisk video in general.
Mihai
On May 15, 2008, at 6:48 AM, Borja SIXTO wrote:
> Hi Mihai,
>
> For the 3G handsets we cannot configure this mode for example (Some
> phones generate always an audio stream).
> So I think it will be great to be able to enable the full VAD with a
> specific application option.
> I will read again the source code.
>
> Thanks,
>
>
> Tech from i6net
>
>
> Mihai Balea a écrit :
>>
>> On May 15, 2008, at 3:53 AM, Borja SIXTO wrote:
>>>
>>> The voice conference works well, but I didn't able to use the video
>>> switching based on VAD or DTMF.
>>
>> Just a quick note here, since this is one of the more confusing
>> aspects of appconference:
>> To get VAD-based video switching working, your clients need to use
>> DTX (i.e. stop sending audio packets when there is no voice
>> activity). While it is possible to implement a full VAD solution in
>> appconference (the relevant code is there), VAD is quite
>> CPU-intensive, so there are advantages to offloading this to the
>> clients.
>>
>> Mihai
>> ------------------------------------------------------------------------
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-video mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-video
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
>
>
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