[asterisk-bugs] [Asterisk 0014927]: [patch] Create option to require audio before reinvite

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=14927 
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Reported By:                BlargMaN
Assigned To:                kpfleming
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Project:                    Asterisk
Issue ID:                   14927
Category:                   Core/NewFeature
Reproducibility:            N/A
Severity:                   tweak
Priority:                   normal
Status:                     ready for review
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-04-17 13:52 CDT
Last Modified:              2009-10-01 11:38 CDT
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Summary:                    [patch] Create option to require audio before
reinvite
Description: 
Due to revision 182527, I can no longer talk to my SIP Gateway, as it
requires audio before a reinvite can be sent.

* no longer sends audio before a reinvite as of the aforementioned
revision, and therefore I needed a way to use/test the 1.6.x branch without
upgrading my current hardware...  

therefore, this patch adds the option 'requireaudio=yes' to
asterisk.conf...

based on the state of this option, asterisk will or will not send audio to
the SIP Gateway before sending a reinvite.
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 (0111729) kpfleming (administrator) - 2009-10-01 11:38
 https://issues.asterisk.org/view.php?id=14927#c111729 
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There is no correlation between audio flowing and re-INVITEs occurring in
any RFC or specification I have seen. It is perfectly reasonable for
chan_sip to issue a re-INVITE of any type (direct media path or T.38)
before any audio has been sent to the other endpoint. If the other endpoint
requires audio before receiving a re-INVITE, then we need to have a
workaround to support them.

With that said, I don't think the supplied patch is the right way to solve
this problem; this is not an Asterisk-wide problem, it is specific to
chan_sip, and it should be solved in chan_sip. In addition, it should be
controllable on a per-peer basis, not only globally. I suggest that the
proper way to control this is for chan_sip to mark each new sip_pvt
(dialog) structure as "no media has passed yet", and then when sip_write()
is called it can mark the dialog as "media has passed". If a re-INVITE is
attempted before audio has passed, and the configuration option to require
media is enabled, then the re-INVITE should be marked as pending (which
chan_sip already has support for), and sip_write() can call
check_pendings() when it marks a dialog as "media has passed", which would
trigger the re-INVITE that is pending. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-10-01 11:38 kpfleming      Note Added: 0111729                          
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