[gst-devel] rtsp / real

[Lutz Müller]

Hello!

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,
payload=(int)101
- rtpdepay1.srcrtcp: no caps
- rtpdepay1.srcrtp: application/x-rtp, media=(string)audio,
payload=(int)101

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
stream? 

gst-launch gstrtspsrc location="rtsp://beagle.unl.edu/relay" !
decodebin ! audioconvert ! alsasink

does not work (no linking in decodebin, probably because of
GST_CAPS_ANY).

Thank you!
-- 
Lutz Müller <lutz at topfrose.de>
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