[asterisk-bugs] [Asterisk 0015484]: [patch] [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:              2011-03-08 16:25 CST
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Summary:                    [patch] [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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---------------------------------------------------------------------- 
 (0132767) murkhog (reporter) - 2011-03-08 16:25
 https://issues.asterisk.org/view.php?id=15484#c132767 
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Some feature ideas that have come up while working with this branch: 

* Ability to leave writestream or readstream empty. A use case: we are
streaming telephone conference calls. This is a broadcast and there is no
RTMP stream coming back into Asterisk. 

I've tried setting the readstream to an unknown stream id ('none') and
this causes a time-out after 5 minutes. The RTMP channel is bridged to a
SIP call to an external provider that connects us to the PSTN. chan_rtmp
does not send any RTP packets for the 'none' stream and I suspect our
provider have the default setting of rtpholdtimeout=300 kick in. 

Anybody know how to send silence from within the chan_rtmp (i. e. how
would the sample data look I write to buffer, how often would I need to do
this)? 

* I'd really like to be able to specify different FMS server, or
application for writestream and readstream. Would it make sense to have
this as part of the channel specification: I. e.
RTMP/server,app,writestream/server,app,readstream. The current settings in
rtmp.conf could still serve as defaults. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-03-08 16:25 murkhog        Note Added: 0132767                          
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