[asterisk-bugs] [Asterisk 0015484]: [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:             1.8
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:              2010-07-25 12:59 CDT
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Summary:                    [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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---------------------------------------------------------------------- 
 (0124986) Harbour (reporter) - 2010-07-25 12:59
 https://issues.asterisk.org/view.php?id=15484#c124986 
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Playing 8KHz speex files with FFPlayback succeeded with clear sound. But
using Dial(RTMP/1/2) always core dumped, last line is :

Jul 25 20:47:28] NOTICE[6464]: chan_rtmp.c:626 rtmp_handle_apacket:
Changed incoming sample rate from 11000 Hz to 8000 Hz
Floating point exception (core dumped)

Core was generated by `asterisk -dcv'.
Program terminated with signal 8, Arithmetic exception.
https://issues.asterisk.org/view.php?id=0  0xb2dd24c1 in av_resample () from
/usr/lib/libavcodec.so.52
(gdb) backtrace
https://issues.asterisk.org/view.php?id=0  0xb2dd24c1 in av_resample () from
/usr/lib/libavcodec.so.52
https://issues.asterisk.org/view.php?id=1  0xffffdc00 in ?? ()
https://issues.asterisk.org/view.php?id=2  0x00000000 in ?? ()

The RTMP server is Wowza 3.0.0, can provide working URL for testing if
needed. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-07-25 12:59 Harbour        Note Added: 0124986                          
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