[asterisk-bugs] [Asterisk 0010868]: 1.4.11 Stable - Polycom phones hang up when media is re-invited while resuming from an on-hold state

[noreply at bugs.digium.com]

The following issue has been REOPENED. 
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http://bugs.digium.com/view.php?id=10868 
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Reported By:                mavince
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   10868
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
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:             10-02-2007 09:23 CDT
Last Modified:              10-05-2007 16:09 CDT
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Summary:                    1.4.11 Stable - Polycom phones hang up when media is
re-invited while resuming from an on-hold state
Description: 
PSTN calls made with Polycom phones (several different firmware loads) hang
up when resuming the call from a hold state. 

The call flow connects a Polycom phone to a PSTN phone through a SONUS
gateway with the media passing directly from to/from the Polycom phone. 

Call Scenario: The call can be initiated and answered in either direction,
to or from the PSTN. RTP is successfully provided. If the Polycom phone end
places the call on hold, MOH will be heard. If the Polycom phone then
attempts to resume the call, Asterisk issues two nearly simultaneous
INVITEs with an incremented CSeq, resulting in a 491 - Request Pending (the
appropriate response). Asterisk acknowledges the 491 and then hangs up the
call! I can reproduce the call behavior consistently.

Key points: media is passing directly to Polycom phone, call can be placed
on-hold successfully, other phones Snom, Aastra work correctly with the
same configuration. When media traverses the Asterisk, a Polycom based call
works correctly.

See Bug Tracker Issue 0009921
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Relationships       ID      Summary
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related to          0009921 1.4.4 sends Re-INVITE twice, resulting ...
related to          0009431 Modify connection: Response 491 not han...
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---------------------------------------------------------------------- 
 mavince - 10-05-07 16:09  
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Significant improvement but not complete.

Replaced stable 1.4.11 rtp.c with modified version. Upon compiling got the
following errors...
[CC] rtp.c -> rtp.o
rtp.c: In function `ast_rtp_read':
rtp.c:1236: warning: passing arg 3 of `ast_sched_add' from incompatible
pointer type
rtp.c: In function `ast_rtp_raw_write':
rtp.c:2654: warning: passing arg 3 of `ast_sched_add' from incompatible
pointer type

Restarted Asterisk and get the following message on every inbound call
-- Incoming call: Got SIP response 500 "CSeq Number Out of order" back
from 172.16.4.26 (an Aastra phone, not involved in the call)

On the positive side, calls originating at the Polycom phones behind
Asterisk (as well as the Aastra) can be alternated between hold and resume
without a call hangup. That's a big improvement.

Calls inbound from the PSTN are answered and can be put on hold. When I
resume the call, the call is disconnected. I have seen behavior where I can
resume once, then go to hold again. When I attempt to resume, the call is
disconnected. 

Issue History 
Date Modified   Username       Field                    Change               
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10-05-07 16:09  mavince        Note Added: 0071565                          
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