[asterisk-bugs] [Asterisk 0016097]: [patch] asterisk crashes when there are no RTP port left

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16097 
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Reported By:                steinwej
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   16097
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     assigned
Target Version:             1.6.0.18
Asterisk Version:           SVN 
JIRA:                       SWP-297 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0 
SVN Revision (number only!): 224449 
Request Review:              
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Date Submitted:             2009-10-19 10:51 CDT
Last Modified:              2009-11-06 11:02 CST
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Summary:                    [patch] asterisk crashes when there are no RTP port
left
Description: 
To reproduce in the lab I changed sip.conf to consume a lot of UDP ports:

videosupport=yes 
t38pt_udptl=yes

rtp.conf:
rtpstart=8000
rtpend=8006

I also configured a sip connection to a different system.
 
Asterisk 1.4:
The first call (SIP ATA - calls SIP trunk) goes through.
The second call is aborted. Asterisk behaves good.
[Oct 19 17:26:24] ERROR[1976]: rtp.c:1965 ast_rtp_new_with_bindaddr: No
RTP ports remaining. Can't setup media stream for this call.
[Oct 19 17:26:24] WARNING[1976]: chan_sip.c:4497 sip_alloc: Unable to
create RTP audio and video session: Address already in use
[Oct 19 17:26:24] ERROR[1976]: chan_sip.c:16076 sip_request_call: Unable
to build sip pvt data for 'sip-test-t38/01229922888w' (Out of memory or
socket error)
[Oct 19 17:26:24] WARNING[1976]: app_dial.c:1183 dial_exec_full: Unable to
create channel of type 'SIP' (cause 42 - Switching equipment congestion)

But asterisk 1.6.0:
First call goes through.
But the second call crashes the system.

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---------------------------------------------------------------------- 
 (0113321) svnbot (reporter) - 2009-11-06 11:02
 https://issues.asterisk.org/view.php?id=16097#c113321 
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Repository: asterisk
Revision: 228415

U   branches/1.6.0/channels/chan_sip.c

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r228415 | file | 2009-11-06 11:02:13 -0600 (Fri, 06 Nov 2009) | 7 lines

Fix a crash caused by freeing a dialog directly instead of using
dialog_unref.

(closes issue https://issues.asterisk.org/view.php?id=16097)
Reported by: steinwej
Patches:
      no_RTP.diff uploaded by steinwej (license 841)

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http://svn.digium.com/view/asterisk?view=rev&revision=228415 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-11-06 11:02 svnbot         Checkin                                      
2009-11-06 11:02 svnbot         Note Added: 0113321                          
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