On 8/31/05, Federico Giannici <giannici@neomedia.it> wrote:

I'd like to ask to somebody with more knowledge of me if a possible
solution to NAT traversal is really feasible.

For various reasons, we DON'T want to use an RTP proxy.
We'd like to avoid the use of STUN because: 1) creates hairpin problems;
2) many UAC have a bad STUN client code implementation; 3) it requires
additional configuration by the final user.

It seems to me that with the nathelper's message rewriting functions it
is possible to solve every problem for the SIP protocol.

Moreover, as we have the REAL IP of the UA (in the original SIP
messages) we could also avoid haipin problems: it is sufficient to use
the original IP/Port of the two UAs if both have the same natted IP (ie
they are behind the same NAT). This doesn't work when the UAs are behind
multiple NATs, but this is a relatively uncommon case.

So, the unresolved problem is with the RTP data, because we don't know
what will be the NATted port so we cannot correctly mangle the SDP data
in the INVITE message.

Am I correct up to this point?

Now, I'm asking myself if it is feasible to use a "MINI RTP Proxy" that
receives the initial INVITEs, discovering the NATted RTP ports, and then
IMMEDIATLY RE-INVITE the two UAs to connect directly each other. So only
the first RTP packet is actually proxed, all subsequent traffic will be
directly between the two UAs.
I think that something similar is done by Asterisk.

Is this feasible?

If it is, then we could have a good solution to NAT Traversal:
1) No Hairpin problems (for one NAT cases)
2) No problems of the normal RTP proxy (waste of bandwidth, longer
delays, bad scalability).
3) Will work with all type of NATs except for symmetric ones (the same
that work with STUN).
4) Simpler UAC configuraton: only username, password and sip server.

Thanks.

--
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    |-                      giannici@neomedia.it
    |ederico Giannici       http://www.neomedia.it
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