[gst-devel] Passing video/audio stream to another RTP library

[Olivier Crête]

On Thu, 2009-01-08 at 11:39 -0700, Merrick Fonnesbeck wrote:
> I have a pipeline that currently streams video using GStreamer to
> another location forming a simple internet call application.
> 
> gst-launch v4l2src !
> video/x-raw-yuv,width=176,height=144,framerate=8/1 ! hantro4200enc !
> rtph263pay ! udpsink host=<ip address> port=<port>
> 
> I want to use SIP to coordinate session information with the
> connection to the destination and I have a SIP library framework that
> I already own a license for that I would like to use (yes I know that
> the N810 come with Sofia-SIP), and this SIP library also comes with
> RTP capabilities.  I am wondering if it is possible for GStreamer to
> pass it's information off to this other library of code and let it
> take care of the RTP transport of streaming video data to the
> destination location and receive incoming data and pass it along into
> GStreamer's own elements for processing and displaying on the screen?
> If anyone knows or has any ideas, please let me know.  Thanks.

Use appsrc and appsink. If you are on a N810, you probably want to
backport the version that have just been merged into gst-plugins-base.

-- 
Olivier Crête
olivier.crete at collabora.co.uk
Collabora Ltd
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