[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:                   minor
Priority:                   normal
Status:                     assigned
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:             2009-07-10 07:30 CDT
Last Modified:              2009-07-15 05:42 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 things :
- autoconf, in version 2.60 or higher
- automake, in version 1.9 or higher
- libavcodec, included in FFMPEG version 0.0.5. Be careful to configure
FFMPEG's sources with the --enable-shared option activated in the configure
script.

Installation procedure :
# svn co http://svn.digium.com/svn/asterisk/team/phsultan/rtmp-support
asterisk-rtmp
# cd asterisk-rtmp
# ./boostrap.sh
[this will generate a new configure script]
# ./configure
# make menuselect
[check here that chan_rtmp is eligible for installation]
# make
# make install
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 (0107768) dlogan (reporter) - 2009-07-15 05:42
 https://issues.asterisk.org/view.php?id=15484#c107768 
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Can you advise where FFMPEG 0.0.5 src can be downloaded from - the site
only seems to have links to V0.5 and trunk.

I have tried compiling against 0.5 - but there are warnings at compile
time and Asterisk seg faults. 

== Spawn extension (default, 500, 1) exited non-zero on
'SIP/snom-00d614d8'
  -- Executing [1002 at default:1] Dial("SIP/snom-00d614d8",
"RTMP/writestream/readstream") in new stack
Segmentation fault

This is a very exciting channel - I am very keen to get this built and
working. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-07-15 05:42 dlogan         Note Added: 0107768                          
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