Can't play back RTP stream (blank) but RTP packets are there.
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.
Can you update the question with the output of
wireshark -vorHelp->About Wireshark:Wireshark.And if you could share a sample capture file (via a file share site. Not a screenshot) that would be nice too.
It is perfectly valid to have SIP traffic from A to B on port 5060 and agree to do RTP from C to D. The main point is here that the RTP stream must match the exact set of IP addesses and ports that SIP agreed upon.
If NAT happen somewehere on the line then you could be in a heap of trouble depending o wht the NAT device will do with ports and the port information of the SIP traffic.
In order to investigate you should measure on all relevant endpoints of IP connnections. Which in your case wouuld be 3 if I read your information correctly.