Posted: Wed May 17, 2006 12:11 am Post subject: [Asterisk-video] RE: AVTF - What's going on?
Quote:
- GStreamer & Asterisk:
Modules for GStreamer (www.gstreamer.net) which allow it to generate
Asterisk format h263 files. With a GStreamer command line you can
convert video files from other formats into h263 and wav files for
playback using Asterisk. I believe that the modules are now
maintained in the GStreamer CVS.
Which H263 rtp format, h263-1996, h263-1998 or h263-2000?
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Posted: Wed May 17, 2006 12:47 am Post subject: [Asterisk-video] RE: AVTF - What's going on?
Hi,
Quote:
> - Videoswitch: (thanks to support from AuPix)
Would it be possible to take this code to app_meetme?
At the time app_conference was an easier starting point. I'll take a
look at app_meetme and see how much work it would be.
Quote:
> - Fast Updates / RTCP / etc etc:
Can we get those. I'll happily open a branch for it and make sure it is
moved forward.
Any feedback on the current rtcp branch/patch?
Most of this is covered by the rtcpbranch you have (and the work by John
Martin) - I've switched to using that code now and will file
bugs/patches as they come up.
Quote:
> - GStreamer & Asterisk:
That is really cool. I'll look at it. Does it divide the video into one
audio and one video file?
Posted: Wed May 17, 2006 4:19 am Post subject: [Asterisk-video] RE: AVTF - What's going on?
Neil Stratford wrote:
Quote:
Hi everyone,
It's great to see so much activity around video.
Here are some projects that we've been working on:
- Videoswitch: (thanks to support from AuPix)
A multi-party video conference module based on app_conference. Each
participant gets to see a single video stream at a time, which can be
switched using either DTMF input from the user or Manager interface
commands. You can use it for things like video conferencing, distance
learning and video broadcast. There is usually a test version running
at 400@sip.vipadia.com if you would like to play with it. We'll be
making the code available soon.
That's awesome, Neil!
We (the original authors of app_conference) were planning on doing this
as well, in the next couple of months; We'll be very happy to see your
code, and work with you to integrate this.
Posted: Wed May 17, 2006 6:56 am Post subject: [Asterisk-video] RE: AVTF - What's going on?
Niel, where can I get a copy of your video app conference? Your work
in general sounds really nice.
----
Shidan
On 5/17/06, Neil Stratford <neils@vipadia.com> wrote:
Quote:
Hi,
>> - Videoswitch: (thanks to support from AuPix)
> Would it be possible to take this code to app_meetme?
At the time app_conference was an easier starting point. I'll take a
look at app_meetme and see how much work it would be.
>> - Fast Updates / RTCP / etc etc:
> Can we get those. I'll happily open a branch for it and make sure it is
> moved forward.
> Any feedback on the current rtcp branch/patch?
Most of this is covered by the rtcpbranch you have (and the work by John
Martin) - I've switched to using that code now and will file
bugs/patches as they come up.
>> - GStreamer & Asterisk:
> That is really cool. I'll look at it. Does it divide the video into one
> audio and one video file?
--
Neil Stratford - http://www.vipadia.com/ - sip:call@vipadia.com
Vipadia Limited :: VoIP Research and Development
_______________________________________________
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Posted: Thu May 18, 2006 3:10 am Post subject: [Asterisk-video] RE: AVTF - What's going on?
Steve Kann wrote:
Quote:
Neil Stratford wrote:
> - Videoswitch: (thanks to support from AuPix)
Quote:
We (the original authors of app_conference) were planning on doing this
as well, in the next couple of months; We'll be very happy to see your
code, and work with you to integrate this.
Hi Steve, that's great - I'll contact you off list.
Neil
--
Neil Stratford - http://www.vipadia.com/ - sip:call@vipadia.com
Vipadia Limited :: VoIP Research and Development
My example uses a input file that's been encoded with different
audio/video codecs but the same principle applies. I believe that there
is an error in the original GStreamer invocation - "rtp263enc" should be
"rtph263pay". GStreamer 0.10 is required!
Posted: Thu May 18, 2006 9:58 pm Post subject: [Asterisk-video] RE: AVTF - What's going on?
Hi everybody,
I see that GStreamer allows encoding video media for Asterisk.
But how to stream a h263 file with Asterisk?
I applied a patch for res_agi.c
(http://bugs.digium.com/view.php?id=5392&nbn=10) to allow the
STREAM FILE AGI command to also stream a video file but this
seems not to work. Someone can help me?
Pierre-Antoine
Quote:
> >> - GStreamer & Asterisk:
> > That is really cool. I'll look at it. Does it divide the
video into one
Quote:
> > audio and one video file?
>
> Yes - you invoke it with a command line like:
>
> gst-launch filesrc location=avin.ogg ! oggdemux name=demux
! theoradec !
Posted: Thu May 18, 2006 10:19 pm Post subject: [Asterisk-video] RE: AVTF - What's going on?
19 maj 2006 kl. 08.58 skrev Pierre-Antoine Strievi:
Quote:
Hi everybody,
I see that GStreamer allows encoding video media for Asterisk.
But how to stream a h263 file with Asterisk?
I don't know about AGI, but there are fishy things here. Video does not
include voice. Playback() assumes that if you have a video file with the
same name as the *audio* file you give as an argument, we will send
that too.
You can't specifiy video format, since we do not transcode anyway.
I guess we have to implement a function that checks if we have video
that is associated with an audio file, like
ISVIDEOSTREAM(filename)
Sorry for not answering your question, but it might give some hints
on how
to solve it.
My example uses a input file that's been encoded with different
audio/video codecs but the same principle applies. I believe that there
is an error in the original GStreamer invocation - "rtp263enc" should be
"rtph263pay". GStreamer 0.10 is required!
Thanks Jeff - you are right - rtph263enc got renamed to rtph263pay. My
mistake!
Neil
--
Neil Stratford - http://www.vipadia.com/ - sip:call@vipadia.com
Vipadia Limited :: VoIP Research and Development
Posted: Fri May 19, 2006 12:12 am Post subject: [Asterisk-video] RE: AVTF - What's going on?
19 maj 2006 kl. 10.54 skrev Neil Stratford:
Quote:
Jeffrey C. Ollie wrote:
>>>> - GStreamer & Asterisk:
>>>
> gst-launch filesrc location=AstriDevCon_Europe_2006.mov ! qtdemux
> name=demux ! ffdec_h263 ! videoscale ! video/x-raw-
> yuv,width=352,height=288 ! ffenc_h263 rtp-payload-size=512 !
> rtph263pay ! asteriskh263 ! filesink location=avout.h263 demux. !
> queue ! ffdec_adpcm_ima_qt ! audioconvert ! audioresample ! audio/
> x-raw-int,rate=8000,channels=1 ! wavenc ! filesink location=avout.wav
> My example uses a input file that's been encoded with different
> audio/video codecs but the same principle applies. I believe that
> there
> is an error in the original GStreamer invocation - "rtp263enc"
> should be
> "rtph263pay". GStreamer 0.10 is required!
Thanks Jeff - you are right - rtph263enc got renamed to rtph263pay.
My mistake!
Neil,
With that simple command line I am surprised you could make a mistake...
And people claim that Asterisk has a bad command line.
Posted: Sat May 20, 2006 2:33 am Post subject: [Asterisk-video] RE: AVTF - What's going on?
19 maj 2006 kl. 15.17 skrev Jeffrey C. Ollie:
Quote:
On Fri, 2006-05-19 at 11:12 +0200, Olle E Johansson wrote:
>
> With that simple command line I am surprised you could make a
> mistake...
>
> And people claim that Asterisk has a bad command line.
>
> Guess we should start writing a doc file.
Well, gst-launch is actually more of a language for building graphs of
audio/video processing plugins. I've attached a PNG that shows a
little
more clearly what's going on...
Jeff, thanks! That helped me understand. Hmmm. Guess I need to learn
about gstreamer.
I added a video.txt file in the /doc directory of trunk. If you want
to add or
change, please send me a diff file or simply the text.
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