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

[Merrick Fonnesbeck]

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.

Merrick


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