We are having a strange intermittent RTP issue.
First let me give you the background story:
Asterisk (VoIP software) intermittently does not send audio back to the callers in the meetme conference bridge. If the caller hangs up and calls back sometimes the audio will work and sometimes it does not. We have taken packet captures and reviewed the SIP and SDP, both are correct and you can actually hear the RTP streams in the packet captures.
I'm trying to use wireshark to debug the issue. It seems that the RTP is getting wonky when going from one server to another. The second packet in the RTP stream says 'Payload changed to PT=0'
I have no clue what this means but it stands out because the other working RTP streams do not have this packet. Does anyone know what this means or how I should continue to debug this issue?
