[*] SIP and nat

[Sean Walberg]

If a peer is marked as "nat=yes", does this mean that Asterisk will use the
externip value in the SDP packets when setting up the media path between the
Asterisk server and the peer?  - Yes, testing shows this.

What about a local peer (ie IP phone) for which Asterisk is setting up a
call? Is there a way to have Asterisk use externip in the SDP info so that
the RTP from the phone builds the NAT channel?  It seems that despite the
NAT setting on the peer, Asterisk always sends the real address in SDP.  The
only way around this is canreinvite=no to force native bridging, or to have
the FW fix up the SDP info. So the nat=xxx setting is only for RTP streams
terminating on the * box itself?

Just trying to confirm my experiments before making a fool of myself in
public :)

Tx,

Sean

-- 
Sean Walberg <sean at ertw.com>    http://ertw.com/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20061208/8c7bc887/attachment.html