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

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10647 
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Reported By:                samdell3
Assigned To:                
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Project:                    Asterisk
Issue ID:                   10647
Category:                   Core/RTP
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
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:              
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Date Submitted:             09-05-2007 00:50 CDT
Last Modified:              09-11-2007 19:51 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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---------------------------------------------------------------------- 
 samdell3 - 09-11-07 19:51  
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Update again

With:
canreinvite=yes
directrtpsetup=yes
Dial(SIP/66500504|10|tThHL(100000:50000:50000))

Completes the call setup properly with direct media, and 2-way audio.
It COMPLETELY ignores the dial options that should keep Asterisk in the
audio path. ie none of the dial options work.


With:
canreinvite=yes
directrtpsetup=no
Dial(SIP/66500504|10|tThHL(100000:50000:50000))

A single re-invite is still issued to one leg of the call, breaking the
RTP flow and causing 1-way audio

chan_sip.c transmit_reinvite_with_sdp is always being hit, and a re-invite
is still sent to the peer that initiated the call, when it shouldn't. 

Issue History 
Date Modified   Username       Field                    Change               
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09-11-07 19:51  samdell3       Note Added: 0070363                          
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