[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:             1.10
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:              2010-11-01 19:33 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 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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---------------------------------------------------------------------- 
 (0128532) cmendes0101 (reporter) - 2010-11-01 19:33
 https://issues.asterisk.org/view.php?id=15484#c128532 
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I'm still having trouble getting this to fully work. I tried rebuilding a
couple times but still having the same issue.

[rtmp]
exten => 500,1,Answer()
exten => 500,2,Dial(RTMP/writestream/readstream)

When I have the caller go into this context, I can use the Publisher demo
app to publish audio(mic) to "readstream" and that works fine, but I cannot
get the Publisher demo app to play audio from the writestream (audio from
the caller).

The only thing that I do get during setup is this error:

   [CC] chan_rtmp.c -> chan_rtmp.o
chan_rtmp.c: In function ârtmp_readâ:
chan_rtmp.c:310: warning: the address of âbufâ will always evaluate as
âtrueâ
chan_rtmp.c: In function ârtmp_handle_apacketâ:
chan_rtmp.c:613: warning: âavcodec_decode_audio2â is deprecated
(declared at /usr/local/include/libavcodec/avcodec.h:3390)
   [LD] chan_rtmp.o -> chan_rtmp.so


Could this have to do with my audio problem?

I dont have a FMS but I'll see what I can do, just so I can eliminate if
its red5 or not. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-11-01 19:33 cmendes0101    Note Added: 0128532                          
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