[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:              2011-04-06 06:53 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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---------------------------------------------------------------------- 
 (0133435) murkhog (reporter) - 2011-04-06 06:53
 https://issues.asterisk.org/view.php?id=15484#c133435 
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Just as follow up to the one-way broadcasting of phone calls (for dlogan
and gabole): 

We now have a set-up that works for us. The outgoing stream goes to our
streaming server (FMS) and we read a silent stream from a seperate Wowza
server to avoid the time-outs.

We use Wowza because it supports scheduling of playlists (as live stream).
We schedule a silent FLV to start in the past (so it starts right away) and
then loop it. Asterisk can use this as incoming stream whenever a call is
established with chan_rtmp. 

More information on the Wowza playlist configuration is here:
http://www.wowzamedia.com/forums/content.php?145-Stream-class-example-with-playlists-and-schedules-set-in-smil-file

I've also uploaded our SMIL playlist and silent FLV:
http://jbauer.es/asterisk/

Note that I had to hack the chan_rtmp.c code. We are using different
streaming servers for incoming and outgoing streams. I hard-coded the
incoming streaming URL to our wowza server. If I find a bit of time, I'll
try to create a patch for chan_rtmp.c to allow the configuration of
different streaming servers and application. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-04-06 06:53 murkhog        Note Added: 0133435                          
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