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<DIV><FONT face=Arial size=2>I understand that setting "canreinvite =
yes" in sip.conf allows clients to send RTP streams directly to each
other and this works only if there is no NAT involved. But if I want to use
two SIP phones behind firewalls and using NAT to be able to send RTP packets to
each other bypassing Asterisk, is there a way to do that?</FONT></DIV>
<DIV><FONT face=Arial size=2>Can this be done if I used IAX instead of SIP?
What exactly does the "notransfer" option in iax.conf do?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Thanks in advance.</FONT></DIV>
<DIV><FONT face=Arial size=2>Regards,</FONT></DIV>
<DIV><FONT face=Arial size=2>Mitul</FONT></DIV></BODY></HTML>