[asterisk-bugs] [Asterisk 0015484]: [branch] RTMP support in Asterisk

Asterisk Bug Tracker noreply at bugs.digium.com
Sat Nov 21 23:22:58 CST 2009


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:              2009-11-21 23:22 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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---------------------------------------------------------------------- 
 (0114104) gpulier (reporter) - 2009-11-21 23:22
 https://issues.asterisk.org/view.php?id=15484#c114104 
---------------------------------------------------------------------- 
My issue gets stranger...
the information passed into the DIAL command makes it to ast_request()
uncorrupted.
numsubst is "writestream/readstream" just before calling ast_request.
But as that data shows up in rtmp_request as the void *data parameter it
is junk.

Thoughts? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-11-21 23:22 gpulier        Note Added: 0114104                          
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