title: RTP: recommend strategy in order to achieve fluent audio stream
body: <p>Let me explain what I mean when I say fluent audio stream.  </p>
  
  <p>I have a VOIP application which transfers PCMU encoded audio wrapped in RTP packages through UDP. I already implemented mechanisms which deal with package losses(as suggested in rfc3550).  </p>
  
  <p>The problem is that due to platform limitations(blackberry OS) I need to maintain a constant flow of data i.e. I need to pass X bytes every S milliseconds.<br>
  Because of network delays, undelivered datagrams etc. I can't guarantee that constant data flow so I created a separate thread which compensates the packages which were dropped or delivered late with fake packages("silence").   </p>
  
  <p>So my question is - can anyone suggest a good way to combine the fake packages and the real ones? I realize that adding a fake package automatically increases the lag and maybe I should ignore a real RTP packages after that but as I said this is because of platform limitations and I am willing to make compromises with the quality of the audio and have some additional speech loss.</p>