[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:                       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:              2010-07-23 09:05 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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---------------------------------------------------------------------- 
 (0124904) phsultan (manager) - 2010-07-23 09:05
 https://issues.asterisk.org/view.php?id=15484#c124904 
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I really can't explain the escaped characters into the URLs, sorry. On my
side, video and audio are properly received, in all cases (file, HTTP,
RTMP).

I just fixed the compile warnings and added coded to properly packetize
the H.263 frames in RTP packets. The consequence is that you don't need to
patch your FFMPEG 0.6 installation anymore. And, the app_ffplayback.c code
is very light.

All my testing have been done with X-Lite and Zoiper. X-lite and Zoiper to
test the video and audio playbaycks respectively. Why? In short, X-lite is
buggy on my MAC and has no sound, while Zoiper can't handle video...

Also, the RTP payload for the H.263 codec has to be set to 115 for X-lite
to accept those packets. And the video codec on X-lite is H.263+(1998).

Finally, the call has to be a video call from the beginning (this happens
on X-lite at least, don't know on Eyebeam).

Can you post a SIP debug output from the Asterisk console? An RTP debug
output would be helpful as well, but it seems like recent changes in the
trunk made this functionnality not available anymore.

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
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2010-07-23 09:05 phsultan       Note Added: 0124904                          
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