[asterisk-bugs] [Asterisk 0015484]: [patch] [branch] RTMP support in Asterisk

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15484 
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Reported By:                phsultan
Assigned To:                phsultan
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Project:                    Asterisk
Issue ID:                   15484
Category:                   Channels/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Target Version:             1.10
Asterisk Version:           SVN 
JIRA:                       SWP-1477 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-07-10 07:30 CDT
Last Modified:              2011-04-24 19:16 CDT
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Summary:                    [patch] [branch] RTMP support in Asterisk
Description: 
I created a new branch that implements Adobe's RTMP (Real Time Media
Protocol).

It allows Asterisk to connect as a client to an RTMP media server like
Red5 or FMS (Flash Media Server), and then publish or receive media streams
from such server. I only tested the connection with Red5.

To install the branch, you'll need several libavcodec, included in FFMPEG
version 0.6. Be careful to configure FFMPEG's sources with the
--enable-shared option activated in the configure script.

Prior to install Asterisk, you need to have librtmp on your system.
librtmp is part of the rtmpdump program : http://rtmpdump.mplayerhq.hu/

To install it :
# wget http://rtmpdump.mplayerhq.hu/download/rtmpdump-2.2e.tar.gz [^]
# tar zxvf rtmpdump-2.2e.tar.gz
# cd rtmpdump-2.2e/
# make
# make install

To install Asterisk :
# svn co http://svn.digium.com/svn/asterisk/team/phsultan/rtmp-support
asterisk-rtmp
# cd asterisk-rtmp
# ./configure
# make menuselect
[check here that chan_rtmp is eligible for installation]
# make
# make install
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 (0134064) cmendes0101 (reporter) - 2011-04-24 19:16
 https://issues.asterisk.org/view.php?id=15484#c134064 
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I cant seem to reproduce this error on our test server, but we have a
server running calls using RTMP and it has been running for a few weeks
with less traffic but the traffic has increased to 10 concurrent SIP in the
past few days and is causing asterisk to crash.

This is what the log is showing right before the crash:
[Apr 24 20:00:42] WARNING[24076]: chan_rtmp.c:299 rtmp_hangup: Asked to
hangup channel not connected
[Apr 24 20:00:42] ERROR[24076]: astobj2.c:258 internal_ao2_ref: refcount
-1 on object 0x2aaaec044118
Also have seen this:
[Apr 24 19:06:30] ERROR[5261]: astobj2.c:116 INTERNAL_OBJ: bad magic
number 0x2d152a38 for 0x1f527370
[Apr 24 19:06:30] ERROR[5261]: astobj2.c:116 INTERNAL_OBJ: bad magic
number 0x2d153078 for 0x1f527370

Sometimes it does not display this error and the email from safe_asterisk
shows "Asterisk on SERVER exited on signal 11." Any suggestions on
debugging this further? 

Issue History 
Date Modified    Username       Field                    Change               
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2011-04-24 19:16 cmendes0101    Note Added: 0134064                          
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