[asterisk-bugs] [Asterisk 0010335]: directrtpsetup doesn't work with 'nat=yes' devices

[noreply at bugs.digium.com]

The following issue has been RESOLVED. 
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http://bugs.digium.com/view.php?id=10335 
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Reported By:                adamgundy
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   10335
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     resolved
Asterisk Version:           1.4.9  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 suspended
Fixed in Version:           
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Date Submitted:             07-30-2007 13:33 CDT
Last Modified:              07-30-2007 14:28 CDT
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Summary:                    directrtpsetup doesn't work with 'nat=yes' devices
Description: 
I've been trying to get two copies of OpenWengo to talk to one another
without asterisk in the middle (video kills my bandwidth!). OpenWengo
unfortunately seems to be very unhappy with SIP reinvites, so I was hoping
the new directrtpsetup behavior would be ideal.

it's worth noting that everything works well without reinvites turned on.

unfortunately, OpenWengo also doesn't appear to do STUN for 'custom' SIP
accounts, so it presents LAN IP addresses to asterisk. with 'nat=yes',
asterisk works around this by replying to the source IP address, rather
than the one specified in the SIP message. HOWEVER, asterisk seems to
figure out the RTP IP address/port by waiting for an incoming packet in
'nat=yes' mode, which of course never happens with 'directrtpsetup=yes'.
instead, asterisk sends both SIP clients the LAN IP addresses and port
numbers.

my thought is, depending on the definition of 'nat=yes', we still might be
able to make this work - eg if 'nat=yes' means 'port forwarding, 1:1
mapping, symmetric NAT', then in theory we should be able to force the RTP
IP address(es) to be the same as the SIP IP address, *IF* the SIP message
contains the same (LAN) IP address for both SIP and RTP? other than that,
we need to fail the direct invite...

that would at least allow people with NAT-blind devices to setup port
forwarding and still have asterisk connect them directly.

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 file - 07-30-07 14:28  
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I'm suspending this since it's a feature request and an item that should be
discussed on the asterisk-dev mailing list. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
07-30-07 14:28  file           Status                   new => resolved     
07-30-07 14:28  file           Resolution               open => suspended   
07-30-07 14:28  file           Assigned To               => file            
07-30-07 14:28  file           Note Added: 0068083                          
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