Welcome to part 3 of our SIP debugging with Wireshark.
Last time around, we discovered that our pcap trace had not captured any RTP packets as a result of a SIP re-invite. That means that in today's episode, we will show you how to change the default Asterisk behaviour so that re-invites are not permitted using the canreinvite command. We will also perform a new tcp dump and then get started on debugging our RTP packets using the Wireshark player.
Next time around Mathias will find the statistical tool which can be used to analyse Audio buffers.
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