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<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">Hash: SHA1</p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">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. </p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">What reason behind this strange behavior is still unknown to us.</p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">Thanks in advance.</p>
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