[asterisk-bugs] [Asterisk 0015285]: [patch] reworked chan_ooh323

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15285 
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Reported By:                may213
Assigned To:                russell
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Project:                    Asterisk
Issue ID:                   15285
Category:                   Addons/chan_ooh323
Reproducibility:            random
Severity:                   feature
Priority:                   normal
Status:                     assigned
Target Version:             1.6.x Version Tracker
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-06-07 18:51 CDT
Last Modified:              2009-07-31 01:09 CDT
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Summary:                    [patch] reworked chan_ooh323
Description: 
There is patch to asterisk-addons-1.6.1.0 for reworked version of 
chan_ooh323 channel module.

main subject of changes is performance and scalability improvement.
one processing thread is replaced with many threads (one for cmdChannel
of EndPoint, one for incoming call creation, one for call processing).

chan_ooh323 with this reworking is more stable and have good performance
in compare with chan_h323.
chan_h323 have memory leaks in all my asterisk systems (which have
versions from 1.2 up to 1.6.1) and grow more than 2Gb virtual memory after
proccessing about 200000 calls. After many years of solving this trouble i
cancel this work and begin work on chan_ooh323.

asterisk 1.6.1 with this version of chan_ooh323 have uptime 4 days and
proccess about 400 000 calls and have 98Mb virtual memory with 4 active
channels and no memory leak detected by asterisk malloc debug system.
(all modules include OOH323 stack code are compiled with MALLOC_DEBUG
options)
Before testing 1.6.1 i work with 1.4 version and had 8 days uptime
asterisk with more than 1000000 calls processed without leaks.

Also there is small changes for new options - incoming call limit, 
numbering of calls, t35country code and vendor/version identification
(albania code is not liked for me ;))

More detail list of changes/improvements is in Changes-ooh323.eng.

======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0015046 [patch] Data truncated in ooh323_reques...
====================================================================== 

---------------------------------------------------------------------- 
 (0108445) sles (reporter) - 2009-07-31 01:09
 https://issues.asterisk.org/view.php?id=15285#c108445 
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Hello!

There is registration:

GCF|192.168.22.19|asterisk|gateway;
RCF|192.168.22.19:1720|asterisk:h323_ID|gateway|6664_ast;
r
AllRegistrations
RCF|192.168.22.19:1720|asterisk:h323_ID|gateway|6664_ast

Here is debug output (tracelevel=6 produces no more output I posted
before)

asterisk*CLI>  ooh323 set debug
OOH323 Debugging Enabled
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
    -- Executing [3098 at sipphones:1] Dial("SIP/6052-09c94360",
"OOH323/3098") in new stack
---   ooh323_request - data 3098 format 0x100 (g729)
---   ooh323_alloc
+++   ooh323_alloc
---   find_peer "3098"
+++   find_peer "3098"
---   ooh323_new - 3098
+++   h323_new
---   onNewCallCreated 9db9f78: ooh323c_o_3
+++   ooh323_request
---   find_call
---   ooh323_call- 3098
+++   find_call
+++   ooh323_call
setting callid number 6052
    -- Called 3098
 Outgoing call 3098(ooh323c_o_3) - Codec prefs - (g729)
asteriskAdding capabilities to call(outgoing, ooh323c_o_3)
	Adding g729 capability to call(outgoing, ooh323c_o_3)
	Adding g729A capability to call(outgoing, ooh323c_o_3)
	Adding g729B capability to call(outgoing, ooh323c_o_3)
---   configure_local_rtp
+++   configure_local_rtp
+++   onNewCallCreated ooh323c_o_3
---   onCallCleared ooh323c_o_3 
---   find_call
+++   find_call
+++   onCallCleared
---   ooh323_hangup
[Jul 31 11:06:05] ERROR[17404]: chan_ooh323.c:987 ooh323_hangup: No call
to hangup
+++   ooh323_hangup
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/6052-09c94360' status is
'CHANUNAVAIL'
---   ooh323_destroy 
 Destroying 3098
 Destroying ooh323c_o_3
+++   ooh323_destroy
    -- Accepting call from '8023400' to '6051' on channel 0/1, span 1
    -- Executing [6051 at default:1] Dial("DAHDI/1-1", "SIP/6051") in new
stack
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
[Jul 31 11:06:11] WARNING[17406]: app_dial.c:1518 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'DAHDI/1-1' status is 'CHANUNAVAIL'
    -- Channel 0/1, span 1 got hangup request, cause 16
    -- Executing [h at default:1] Goto("DAHDI/1-1", "-CHANUNAVAIL,1") in new
stack
    -- Goto (default,-CHANUNAVAIL,1)
    -- Hungup 'DAHDI/1-1' 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-07-31 01:09 sles           Note Added: 0108445                          
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