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

[Neil Stratford]

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