how to get balance between rtp socket and appsrc push data for gstreamer processing

[Soho Soho123]

Hi All,

I setup a rtp stream server by gst-launch:
./gst-launch-0.10 -v filesrc location=/home/soho/audio_test/1_48K.wav
! wavparse ! audioconvert ! rtpL16pay ! udpsink host=192.168.1.252
port=8554

And I implement a rtp socket to receive the rtp payload from the
sender that I start.
I just can hear about 3~4 seconds output from my alsa device, then all
of the sounds like noise.
the code that I implement is :
1. create pipeline:
appsrc->decodebin2->audioconvert->audioresample->alsasink
set caps and property for appsrc.
2. call gst_app_src_push_buffer when rtp socket get new payload

the rtp data from rtp stream sender is 7ms every packet, length is
about 1388 bytes.
does anyone have idea how to get balance between rtp socket and
gstreamer processing?

Thanks!
Soho