[OpenSIPS-Users] nat_traversal module

[Iñaki Baz Castillo]

El Martes, 2 de Junio de 2009, Gavin Henry escribió:
> 2009/6/2 Iñaki Baz Castillo <ibc at aliax.net>:
> > El Martes, 2 de Junio de 2009, Gavin Henry escribió:
> >> Hi,
> >>
> >> Does http://www.opensips.org/html/docs/modules/1.5.x/nat_traversal.html
> >> work in tandem with MediaProxy and RTPproxy to handle SIP signalling?
> >
> > Not in tandm, it just works perfectly (it "fixes" NAT issues in
> > signalling while RrtpProxy/MediaProxy "fix" NAT issue related to media.
> >
> > --
> > Iñaki Baz Castillo <ibc at aliax.net>
>
> Yes, I should have read the docs:
>
> "The nat_traversal module provides support for handling far-end NAT
> traversal for SIP signaling. "
>
> I'm confused, wouldn't you need to do both is SIP signalling is
> suffering from NAT issues? Or does it depend on media routes?

You need both.

nat_traversal will fix the NAT issues in signaling (making possible requests 
in-dialog as re-INVITE, BYE... to arrive to the natted destination, mantaining 
the NAT keepalive for INVITE, REGISTER, SUBSCRIBE...).

mediaproxy or nathelper (rtpproxy) modules allow rewritting the SDP emdia 
address in order to force the RTP through a media proxy (MediaProxy or 
RtpProxy). This will fix the audio issues when caller and/or called are behind 
NAT.




-- 
Iñaki Baz Castillo <ibc at aliax.net>