So it's a problem with my NAT, that's what I thought.. Ok, I got
another question, with SIP nantive brideing happening, do the CDRs
asterisk keeps still good enough for billing, or are they only good for
the short time asterisk is in the media stream?
Kevin P. Fleming wrote:
Nathan Goodwin wrote:
Does anyone have any ideas how I could fix this, this is sort of
important, if it's just me because of my NAT causing it, would doing
so part forwarding and disable NAT support on asterisk and the Sipura
fix this problem?
It's almost impossible to fix this problem. Here's the scenario:
Your SPA-2000 initiates a call to the * server, and then * initiates a
call to your provider. When the provider answers, * tells the
SPA-2000, and it starts sending RTP to *. By doing so, your
NAT/firewall expects to receive packets back from the _same IP address
and port they were sent to_. While * is still in the media path, this
is how it works, and things are fine.
However, when * tries to re-invite the SIP provider to send audio
directly to your SPA-2000, the packets now arrive from a different IP
(and probably a different port number). Any decent NAT/firewall will
drop them on the floor. Thus, no audio from the provider.
It is _possible_ for this to work if * happens to reinvite your
SPA-2000 _first_, and it starts sending audio directly to the SIP
provider, thus opening a different IP/port combination through the
firewall. However, this is not reliable, and there is no way to force
the reinvites to happen in a particular order (or to complete in a
particular amount of time).
You can disallow reinvite for your SPA-2000 but leave it turned on for
your provider, in which case one direction of the media stream can
bypass *. Otherwise, you need a NAT/firewall that understands SIP so
it can be aware of the changes as they occur (or you need to use a SIP
proxy on the NAT/firewall, like siproxd).
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