From: Olivier C. <oli...@co...> - 2009-01-08 18:53:37
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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 oli...@co... Collabora Ltd |