trying to create a video chat app.

[Lab work]

Hi

Thanks for the replay.
Your answer left me a bit confused.

>Finally, each streams should be sent to it's down socket in RTP, you may
multiplex them but this is a lot more work.

Do you mean that I should not mux them and do something like:

pipeline 1:

gst-launch-1.0 v4l2src !  video/x-raw,width=640,height=480 !  timeoverlay !
 tee name="local" !  queue !  autovideosink local. !  queue ! x264enc
tune=zerolatency byte-stream=true bitrate=500 threads=1 ! h264parse
config-interval=1  ! rtph264pay !  udpsink host=192.168.100.3 port= 5000
udpsrc port=5000 caps=\"application/x-rtp,payload=96,encoding-name=H264\" !
queue ! rtph264depay ! h264parse ! decodebin ! autovideosink

pipeline 2 :

gst-launch-1.0 pulsesrc ! audioconvert ! audioresample ! speexenc !
rtpspeexpay ! udpsink host=192.168.100.3 port=4444 udpsrc port=4444
caps="application/x-rtp, media=(string)audio, clock-rate=(int)16000,
encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" !
rtpjitterbuffer ! rtpspeexdepay ! speexdec ! audioconvert ! audioresample !
autoaudiosink


And playing similar pipelines on other PC. I have tried this type of
approach and the video part was working but audio part was creating problem.
when I run the second pipeline there is no audio at either end while they
work fine independently. Can you please explain the reason. And me out of
this problem.
Thank you.

On Mon, Jun 22, 2015 at 7:32 PM, Nicolas Dufresne <
nicolas.dufresne at collabora.com> wrote:

> Le lundi 22 juin 2015 à 17:00 +0530, Lab work a écrit :
> > gst-launch-1.0  oggmux name="muxer"  v4l2src ! video/x-raw,
> > framerate=30/1, width=640, height=480 ! videoconvert ! x264enc !
> > multiqueue ! muxer.  videotestsrc ! video/x-raw, framerate=30/1,
> > width=640, height=480 ! videoconvert ! x264enc ! multiqueue ! muxer.
> > autoaudiosrc ! audioconvert ! speexenc ! queue ! muxer.  udpsink
> > host=127.0.0.1 port=5000
>
> So you want to stream in RTP over UDP. First you need to drop this
> unlinked oggmux. OGG is not RTP. Also it does not work with variable
> framerate (which most Logitech camera produces). Second, pick a preset
> on x264enc that is suited for live (like tune=zerolatency). Finally,
> each streams should be sent to it's down socket in RTP, you may multi
> -plex them but this is a lot more work.
>
> Nicolas
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> gstreamer-devel at lists.freedesktop.org
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>
>
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