I can see all the SIP packets and UDP/RTP packets in the packet list. (G711) Analyze voip calls shows the list of calls. If pick one and hit play back audio, after it processes the file it comes back blank. If I change the display filter to include the UDP port the packets are on and click on a sample packet and hit telephony -> rtp -> rtp player it will add that stream to the RTP player. I have to repeat the process for each updated stream after every re-invite.
One "unusual" thing about these streams is the source audio is not the same IP address as the source SIP invite. The invite header specifies a different IP address for audio.
