VoIP RTP issue

[Vik Killa <vipkilla () gmail com>]

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?

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