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 To :  yate@v...
 From :  "martin A." <acevedoma@h...>
 Subject :  Re: [yate] about nat
 Date :  Mon, 12 Dec 2005 12:29:13 +0000
Hi thank you for your answer... I need a full NAT support rtp and signalling 
using sip or h323, do you know any sip client or stack that support this 
with yate? Or it is impossible to do full NAT with yate?
Thanks ...

>There are two steps involved in NAT traversal:
>
>1. If the endpoint declares a private IP address for voice but it has a
>public signalling address then Yate assumes the client is behind NAT and
>will substitute the signalling address for voice (RTP data). The old
>address is still available in the call.route and call.execute messages as
>"rtp_nat_addr" so this action can be undone per call.
>
>2. Only if RTP data passes trough Yate it will detect when data comes in
>from the wrong source address and will adjust its own destination address
>accordingly. This works very well with SIP but may not work with all H.323
>endpoints as H.323 supports (but is rarely used) asymmetric RTP.
>
>If RTP forwarding is used the data does not pass trough Yate so step 2
>must be accomplished by each endpoint. The public address is still
>declared as RTP address so endpoints should be at least able to find each
>other.
>
>As far as I know the RTP of OpenH323 and OPAL is able to perform 2 but it
>may depend on configuration.
>
>For the call case you described it is important for the endpoint outside
>NAT (the H.323 one) to be able to perform function 2 if RTP is forwarded
>directly between endpoints. That's because the endpoint behind NAT may be
>unaware that the port ahs been translated as well.
>
>
>Paul Chitescu
>
>
>On Sun, 11 Dec 2005, martin A. wrote:
> > I saw yestarday in your site a solution for nat traversal using sip. Is 
>this
> > solution compatible with opal end points(using sip)? and does this 
>solutions
> > works with yate sip-h323 call? , I mean the original call is a sip 
>behind a
> > nat connected with yate and yate with h323 endpoint(not behind a nat) 
>using
> > the signaling proxy or does it only work with sip to sip calls....?.
> >
> > Thanks..