[OpenSIPS-Users] nat_traversal module

[Gavin Henry]

2009/6/2 Iñaki Baz Castillo <ibc at aliax.net>:
> 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.

OK, as I thought. Thanks for the confirmation.

Gavin.

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