Problem Streaming Live Audio Using RTP and UDPSink

[William Metcalf]

I am trying to capture live audio from a capture card in a gstreamer 
application, encode it into AAC format, and then send the audio over a 
network using UDPSink and RTP.  My method of accomplishing this task 
works almost perfectly, except that occasionally the audio will become 
very jumpy for a few seconds and then return to normal, and then after a 
few seconds it will get jumpy, etc.  I am not getting any errors when I 
play the audio, so I am assuming that it must be some property I am not 
setting correctly, or maybe there is an element I am missing which can 
help solve the problem.  My pipelines are as follows:

Server Pipeline: appsrc max-bytes=8000 is-live=true typefind=true ! 
audioparse ! faac bitrate=320000 ! rtpmp4apay host=192.168.42.68 port=52222

Client Pipeline: udpsrc port=52222 ! 
"application/x-rtp,media=(string)audio,clock-rate=(int)44100,encoding-name=(string)MP4A-LATM,payload=(int)96" 
! rtpmp4adepay ! 
"audio/mpeg,mpegversion=(int)4,channels=(int)2,rate=(int)44100" ! faad ! 
autoaudiosink

I am very close to having this work, so any help that anyone can provide 
will be greatly appreciated!

William