[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:              2009-12-11 04:53 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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---------------------------------------------------------------------- 
 (0115140) vmanthei (reporter) - 2009-12-11 04:53
 https://issues.asterisk.org/view.php?id=15484#c115140 
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@gpulier

Replace the following lines in the method rtmp_new:

i->encoder = avcodec_find_encoder(CODEC_ID_PCM_S16LE);
by
i->encoder = avcodec_find_encoder(CODEC_ID_ADPCM_SWF);

and

i->encoding_context = avcodec_alloc_context2(CODEC_ID_ADPCM_SWF);
i->encoding_context->channels = 1;
i->encoding_context->sample_rate = 11000;
by
i->encoding_context = avcodec_alloc_context2(CODEC_TYPE_AUDIO);
i->encoding_context->codec_id = CODEC_ID_ADPCM_SWF;
i->encoding_context->codec_type = CODEC_TYPE_AUDIO;
i->encoding_context->channels = 1;
i->encoding_context->sample_rate = 11025;

Works fine for me. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-12-11 04:53 vmanthei       Note Added: 0115140                          
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