[Asterisk-Users] SIP native bridge problem

Nathan Goodwin ngoodwin at nycap.rr.com
Sat Jan 29 23:30:16 MST 2005


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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