[Asterisk-Dev] Asterisk SIP behind NAT. Possible solution.

[Jamie Carl]

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Hey guyz,

I've noticed that most proxies out there these days are actually
implementing the VIA headers properly and returning a "received="
option, containing the IP address a msg was received from.  Is it
possible to store this and use it for the RTP stream destination to
solve this SIP behind NAT there's a few of us having?

At the moment the SDP section transmits the internal IP address of the
asterisk server which means the remote UA sends RTP packets to the wrong
IP.

So how about in sip.conf, something like:

usereceived=yes

And when registering, when asterisk receives the OK message with the VIA
header:

Via: SIP/2.0/UDP
10.50.1.2:5060;received=203.51.67.9;branch=z9hG4bK23f994c2

using the received option as the destination for RTP packets in the SDP
section.  

Would this work?

Regards,

Jamie Carl
Email:	me at jazz-inc.net
PH:		+61-414-365-466