[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:                        
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:              2009-11-21 06:18 CST
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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 0.5. 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
# ./configure
# make menuselect
[check here that chan_rtmp is eligible for installation]
# make
# make install
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 (0114095) phsultan (manager) - 2009-11-21 06:18
 https://issues.asterisk.org/view.php?id=15484#c114095 
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The code needs to be updated, at least to reflect changes in the core
Asterisk API, as nychip spotted it in note
https://issues.asterisk.org/view.php?id=113710, and also to solve the
inability to set up the socket to the RTMP server.

I'll review the code and adapt it by the end of next week. Also, I don't
have any FMS platform here to test with, so I suggest we focus on making it
work with Red5 first.

I was also thinking about an improvement that could allow us to stop using
FFMPEG. I read somewhere that Flash clients (version 10 or higher) can use
Speex. If that's the case, then we can make them talk directly with
Asterisk's rather than transcode the audio packets (codec : Nellymoser
ASAO) with FFMPEG. That needs to be investigated.

Thanks everybody for all the testing and feedback! 

Issue History 
Date Modified    Username       Field                    Change               
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2009-11-21 06:18 phsultan       Note Added: 0114095                          
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