AFAIK, two UAs (symm) behind two different port restricted cone NATs
can talk to each
other without the mediaproxy, try to fix the SDP using fix_nated_sdp("2").
If the NAT is hairpin enabled then UAs behind the same port restricted NAT
can talk to each other.
~Vamsi
On 9/25/06, kjcsb <kjcsb(a)orcon.net.nz> wrote:
Yes, you are most definitely on to something.
NAT-handling is complex
and
it takes some work to fine-tune it the way you
want. I few comments:
- Look at nathelper's nat_uac_test. It has more options and better
control, look at option 16, which is very good for detecting symmetric
NATs where STUN or an ALG has tried to fix the message
- If you are doing pstn, your gw supporting active media will reduce
your
proxied calls to none
- sipura has many nat-handling options and takes some tweaking to get
them
right for your config
- The behavior of the UAs will differ depending on the type of NAT they
are behind. When behind a symmetric NAT, they should not try to fix the
ip:port, but some do. nat_uac_test("16") will in most cases reveal this
Good luck! (and I'm sure others would appreciate a how-to on optimizing
NAT at
iptel.org
http://www.iptel.org/node/add/flexinode-4
If you create one, I'll help out in making it accurate)
Also, make sure you have a look at the new NAT-handling document:
http://www.iptel.org/ser/howtos/optimizing_the_use_of_rtp_proxy
g-)
Many thanks. I've read and reread "Optimizing the use of rtp proxy".
I've
also done a lot more reading on SDP & RTP which is most relevant to the
audio issue. Signalling is not the problem i.e. the messages are passed
back
and forward through the proxy and I'm happy with that. It's the audio I
want
to offload.
I think the key unanswered question I have is this: in the (seemingly)
most
common scenario of two symmetric (signalling and RTP) UAs behind two
different (port) restricted cone NATs, can two-way audio be established
without the use of a media proxy? I had previously thought that was
possible
but the latest reading I have done indicates not. Why? Because one side
must
initiate the audio part of the call and the other side's NAT device will
not
know where to send that audio on the LAN side of the network. Could
someone
put me out of my misery and confirm one way or the other?
I had thought another alternative was to map the RTP ports on the NAT
device. This would mean forwarding ranges of ports to specific IP
addresses
(each different port range relating to a specific UA) on the NAT device.
Each UA would then be configured to send RTP traffic on the port range
relating to its IP address. But if both sides are behind NAT then am I
right
in thinking that this won't work either because the callees NAT device
still
doesn't know where to send it?
Regarding me documenting my solution it looks to me like it's already been
done in "Optimizing the use of rtp proxy"! I'm currently using media proxy
so the main difference would be that the media proxy selection would be
based on the domain rather than an avp.e.g.
west.domain.com goes to one
proxy and
east.domain.com goes to another.
Cameron
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