[asterisk-bugs] [Asterisk 0018485]: [patch] IAX2 Retry Time Review

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18485 
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Reported By:                netfuse
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18485
Category:                   Channels/chan_iax2
Reproducibility:            have not tried
Severity:                   tweak
Priority:                   normal
Status:                     ready for review
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-12-16 09:41 CST
Last Modified:              2010-12-16 11:10 CST
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Summary:                    [patch] IAX2 Retry Time Review
Description: 
Guys,

The IAX2 channel driver has a number of timeout settings for
communication. I have found that customers with network problems are not
particularly benefitted by these timeouts because they are so incredibly
high.

If you look at the maths, it will be at least 30 seconds with absolutely
no traffic before a DIAL on an IAX2 host fails. However, a SIP peer with
qualify=yes defaults to 2000ms. I recommend that the 2 seconds timeout
should be applied to dialling on IAX trunks also.

Patch attached for your consideration. I have tested this in production
use with good results.

Cheers
Leo
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 (0129688) netfuse (reporter) - 2010-12-16 11:10
 https://issues.asterisk.org/view.php?id=18485#c129688 
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Sorry, I guess you misunderstand. My point is that latency over 2 seconds
is clearly not suitable for general communication - hence why 2000ms is the
default qualify value for SIP.

By this thinking, we should see similar timeouts on other technologies.

You could argue that a custom (non-Asterisk) IAX server would defer a call
acceptance until it'd done some background checks (looked up the Caller ID,
say) but this should still send an ACK and this would prevent the Dial()
being aborted.

Adding these reduced timeouts means that you can have meaningful route
prioritisation. Take the example of FreePBX "Outbound Routes". You can add
as many routes as you want and set priority, but if the first one can not
be reached, it will be a long time before the next route is attempted.
These modified values suddenly make that route order useful again!

Anyway, I'm sure someone will disagree with my values, but we've used the
first version (500ms and 1 retry) in production for a good while and it
works well. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-12-16 11:10 netfuse        Note Added: 0129688                          
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