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

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Dec 10 20:39:27 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:              2009-12-10 20:39 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
====================================================================== 

---------------------------------------------------------------------- 
 (0115135) gpulier (reporter) - 2009-12-10 20:39
 https://issues.asterisk.org/view.php?id=15484#c115135 
---------------------------------------------------------------------- 
Thanks for the new build, it seems to get a lot further for me!

My next issue arises when Asterisk gets to this line in my
extensions.conf:
exten => _XX.,n,Dial(RTMP/writestream/readstream)

when it gets to this point, Asterisk running in debug mode shows:
[pcm_s16le @ 0x9aa1b60]codec type or id mismatches
Segmentation fault

The good news is that I do see a connect and disconnect in my red5 logs,
so it seems like its almost working.

My ffmpeg info looks like this:
FFmpeg version SVN-r20557, Copyright (c) 2000-2009 Fabrice Bellard, et
al.
  built on Nov 20 2009 13:50:55 with gcc 4.3.2
  configuration: --enable-shared --disable-optimizations --disable-mmx
  libavutil     50. 4. 0 / 50. 4. 0
  libavcodec    52.41. 0 / 52.41. 0
  libavformat   52.39. 2 / 52.39. 2
  libavdevice   52. 2. 0 / 52. 2. 0
  libswscale     0. 7. 1 /  0. 7. 1

Any suggestions?
Any suggestions? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-12-10 20:39 gpulier        Note Added: 0115135                          
======================================================================




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