I have spend the last week working on the MGCP Gateway of OpenBSC to allocate the network and the BSC/BTS port separately, to add a feature to forward the RTP stream from the BTS IN/OUT, NET IN/OUT to another system. On this different system one can use something like GStreamer to decode the stream and listen to it. This can be useful to debug when the voice doesn't arrive where it should.
In a network simulation with Linux's netem we have tried to simulate a bad vsat link and wanted to see how big the latency/jitter can be to still have an acceptable voice call and now I will play a bit with RTP jitterbuffers... This allows me to look at GStreamer once again to see if their jitter buffer is finally working.