Need help with using OPUS over RTP

[envion]

Thank you for the answer. I'm now able to send audiotestsrc with this set up.


Client:  gst-launch-0.10 audiotestsrc ! audioconvert ! audioresample !
opusenc ! rtpopuspay ! udpsink host=reciver port=5000 


Reciver: gst-launch-1.0 udpsrc caps=$AUDIO_CAPS port=5000 ! rtpjitterbuffer
latency=20 do-lost=True ! rtpopusdepay ! opusdec plc=true ! alsasink

Where
AUDIO_CAPS=="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)X-GST-OPUS-DRAFT-SPITTKA-00"


When I change audiotestsrc to pulsesrc I don't get any audio out from the
Reciver. Do I need to add more things to the pipeline when I'm handling
pulsesrc ?

The command that don't work at the server is this one (just to be clear):
gst-launch-0.10 pulsesrc ! audioconvert ! audioresample ! opusenc !
rtpopuspay ! udpsink host=reciver port=5000 



Thanks again
Alexander



--
View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Need-help-with-using-OPUS-over-RTP-tp4661409p4661541.html
Sent from the GStreamer-devel mailing list archive at Nabble.com.