[Asterisk-video] RE: AVTF - What's going on?

[Shidan]

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 at vipadia.com> wrote:
> 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?
>
> Yes - you invoke it with a command line like:
>
> gst-launch filesrc location=avin.ogg ! oggdemux name=demux ! theoradec !
> ffmpegcolorspace ! videoscale ! video/x-raw-yuv, width=352, height=288 !
> ffenc_h263 rtp-payload-size=512 ! rtph263enc ! asteriskh263 ! filesink
> location=avout.h263 demux. ! queue ! vorbisdec ! audioconvert !
> audio/x-raw-int ! audioconvert ! wavenc ! filesink location=soxin.wav &&
> sox soxin.wav -r 8000 -c 1 -s -w avout.wav resample -ql
>
> --
> Neil Stratford - http://www.vipadia.com/ - sip:call at vipadia.com
> Vipadia Limited :: VoIP Research and Development
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