[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:              2009-07-16 10:53 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 0.5. 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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 (0107848) phsultan (manager) - 2009-07-16 10:53
 https://issues.asterisk.org/view.php?id=15484#c107848 
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I've been testing the Publisher demo application from Red5 for my tests.
You should set the Buffer value to 0 under the Server tag, and the Timeout
value to 0 under the Audio tag. I think we can check why Asterisk does not
connect to FMS later (I know there are subtle differences in the connection
handling process between FMS and Red5).

After that you can play the stream Asterisk is publishing, and publish
back a stream named of your choice that will be read by Asterisk.

I think the fact that you're running this code on a 64 bits computer might
cause problems, and the endianness certainly will as well. Just for
information, I'm running a 32 bits / little endian computer.

I've just committed code that should fix the Asterisk related warnings.
However, I can't really explain why the deprecation message for the
avcodec_decode_audio2 does not show up on my computer.

Also, I committed the configure file, so you won't need to run the
bootstrap.sh script before configure.

Finally, you're right, if we properly implement Speex pass-through in this
code, we won't need FFMPEG to resample or transcode the media packets.
That's something we need to add!

Thanks a lot for your valuable input. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-07-16 10:53 phsultan       Note Added: 0107848                          
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