[gst-devel] rtsp / real
lutz at topfrose.de
Fri Jul 21 23:27:20 CEST 2006
I ported the handshake logic between the client and a Real server from
xine to gst-plugins-good/gst/rtsp.
I now get 4 pads:
- rtpdepay0.srcrtcp: no caps
- rtpdepay0.srcrtp: application/x-rtp, media=(string)video,
- rtpdepay1.srcrtcp: no caps
- rtpdepay1.srcrtp: application/x-rtp, media=(string)audio,
Can someone tell me what elements I need to connect those pads to? It
would help a lot if I had a sample working rtsp pipeline, i.e. for
rtsp://beagle.unl.edu/relay. How should the pipeline be for this
gst-launch gstrtspsrc location="rtsp://beagle.unl.edu/relay" !
decodebin ! audioconvert ! alsasink
does not work (no linking in decodebin, probably because of
Lutz Müller <lutz at topfrose.de>
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