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

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A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10868 
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Reported By:                mavince
Assigned To:                
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Project:                    Asterisk
Issue ID:                   10868
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
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:              10-04-2007 16:51 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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 andrew - 10-04-07 16:51  
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I just tried this quickly and had no problems (using asterisk branch 1.4).

How do you have your "hold" setup on the polycom phone? (it's in the phone
config file). Polycom has two types of "hold".

In my SIP.CONF setup I have for each device:
NAT=NO
CANREINVITE=YES

My test phones: Polycom 650 and 600 with 2.2.0 and 2.1.2 firmware
My test gateway: Cisco AS5300 with IOS 12.3
Two firewalls between the phones and asterisk/cisco (but no NAT used)

RTP media stream is going directly between the two phones, or directly
between the polycom and the cisco. It is not using asterisk, as verified by
tcpdump (but see more info).

I used hold and resume on each phone several times. When on hold the
stream to the polycom stopped (as expected) and the other end had hold
music. When I resumed off hold I had two way audio (as expected).

Now for the strange part. When "canreinvite=yes" I got an asymmetric RTP
media stream between the polycom/asterisk/cisco. The polycom would send
data directly to the cisco but the cisco sends data to asterisk (and then
to the phone). After a hold/resume the media data is direct between the
phone and the cisco gateway (asterisk is no longer in the
middle)....strange... seems something may be off with the first invite with
the cisco. It did not matter which side started the call. 

Issue History 
Date Modified   Username       Field                    Change               
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10-04-07 16:51  andrew         Note Added: 0071485                          
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