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

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Nov 16 13:20:51 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:              2009-11-16 13:20 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
====================================================================== 

---------------------------------------------------------------------- 
 (0113876) vhannon (reporter) - 2009-11-16 13:20
 https://issues.asterisk.org/view.php?id=15484#c113876 
---------------------------------------------------------------------- 
Thank nychip.

Now that it compiles we're having a little issue connecting to the FMS
server.  We get this error when we make a call.  

-- Executing [1234 at Encode:2] Dial("SIP/voicepulse-primary-00000004",
"RTMP/writestream/null") in new stack
[Nov 15 01:37:11] WARNING[9807]: chan_rtmp.c:682 rtmp_request: The RTMP
driver requires a stream identifier to read
[Nov 15 01:37:11] WARNING[9807]: app_dial.c:1818 dial_exec_full: Unable to
create channel of type 'RTMP' (cause 0 - Unknown)

The basic dialplan is to answer and forward to the extension that calls
rtmp:
exten => _XX.,n,Answer
exten => _XX.,n,Playback(hello-world)
exten => _XX.,n,Goto(Encode,1234,1)


[Encode] ;Send call to FMS
exten => 1234,1,Playback(hello-world)
exten => 1234,n,Dial(RTMP/writestream/null)

(ftr, both Hello world audio files play)
Thanks for any help. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-11-16 13:20 vhannon        Note Added: 0113876                          
======================================================================




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