We use a obfuscation software to encrypt/mangle both SIP/RTP which sits
before asterisk. What happens is sometimes we don’t get any voice. after
some “rtp set debug” we found out that when received ip of the rtp stream
is router’s public ip, everything works cleanly. But sometimes we get the
private ip’s of the client as received address in rtp stream which results
in “no voice”. it seems asterisk because of some unknown reason failed to
traverse nat for the media stream.
What reason behind this strange behavior is still unknown to us.
Thanks in advance.