[asterisk-bugs] [Asterisk 0010647]: SIP Reinvite behaviour does not work as expected with certain dial() options

[noreply at bugs.digium.com]

The following issue has been RESOLVED. 
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http://bugs.digium.com/view.php?id=10647 
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Reported By:                samdell3
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   10647
Category:                   Core/RTP
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     resolved
Asterisk Version:           1.4.11  
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:             09-05-2007 00:50 CDT
Last Modified:              05-05-2008 09:45 CDT
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Summary:                    SIP Reinvite behaviour does not work as expected
with certain dial() options
Description: 
With canreinvite=no on any SIP peer, the call is never reinvited (this is
correct behaviour)

With canreinvite=yes, and using with a Dial() option from below,
re-invites are not issued correctly. (actually, reinvites should not be
issued at all...)

Asterisk is not supposed to perform a re-invite when using any of the
following Dial() options: t, T, h, H, w, W or L (with multiple arguments)
This is not the case. 
Asterisk still issues a Re-invite to one of the call legs causing an
asytmetrical RTP traffic flow (causing one-way audio if the SIP peer
filters RTP packets coming from somehwere that was not in it's own SDP)

EG 
SIPPeerA------ASTERISK-----SIPPeerB

SIPPeerA calls SIPPeerB

If either or both SIPPeerA or SIPPeerB have canreinvite=no, the RTP flow
is always via Asterisk - this is correct.

If Both SIPPeers are canreinvite=yes, AND the dial command contains any of
the above dial() options, then the RTP flow forms a triangle due to a
single re-invite STILL being issued by Asterisk. EG A's RTP goes to
Asterisk, Asterisk's RTP goes to B, but B's RTP goes to A. This is because
Asterisk issues a re-invite and tells B to talk to A when it shouldn't. 
If Asterisk does issue a re-invite for one leg, it should issue a
re-invite for both legs! But in this case it should not issues any
re-invites at all.
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---------------------------------------------------------------------- 
 file - 05-05-08 09:45  
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While I hate to suspend this it has been quite some time, and I've given
all the information requested. Was the issue outside the scope of Asterisk?
Did you need more information? 

Issue History 
Date Modified   Username       Field                    Change               
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05-05-08 09:45  file           Note Added: 0086407                          
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