Trying to get the rtp payload in buffer with rtpL16depay

[Rohan0993]

Hi all,
I am new to gstreamer. I have successfully made this pipeline - 

*host:* gst-launch-1.0 audiotestsrc ! audioconvert !
audio/x-raw,channels=1,depth=16,width=16,rate=44100 ! rtpL16pay  ! udpsink
host=192.168.151.99 port=5000

*client:* gst-launch-1.0 -v udpsrc port=5000 !
"application/x-rtp,media=(string)audio, clock-rate=(int)44100, width=16,
height=16, encoding-name=(string)L16, encoding-params=(string)1,
channels=(int)1, channel-positions=(int)1, payload=(int)96" ! rtpL16depay !
audioconvert ! autoaudiosink sync=false

I have coded this* same pipeline in C++ as well *which works fine.

Now I want to take the *rtpL16depay output in a buffer.* I have tried the
Gstreamer element *appsink* like this (and it do not give any pipeline
error) -

*client:* gst-launch-1.0 -v udpsrc port=5000 !
"application/x-rtp,media=(string)audio, clock-rate=(int)44100, width=16,
height=16, encoding-name=(string)L16, encoding-params=(string)1,
channels=(int)1, channel-positions=(int)1, payload=(int)96" ! rtpL16depay !
appsink

What does this above command (with appsink) do? Where is the buffer saved
(if it did)? How can I access this buffer? /Please note the comands on
terminal are for testing whether the pipeline works, the code is then being
written in C++./



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