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

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Dec 11 13:25:27 CST 2009


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=15484 
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Reported By:                phsultan
Assigned To:                phsultan
====================================================================== 
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:              
====================================================================== 
Date Submitted:             2009-07-10 07:30 CDT
Last Modified:              2009-12-11 13:25 CST
====================================================================== 
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
====================================================================== 

---------------------------------------------------------------------- 
 (0115154) gpulier (reporter) - 2009-12-11 13:25
 https://issues.asterisk.org/view.php?id=15484#c115154 
---------------------------------------------------------------------- 
More specifically, my setup now enters the function rtmp_alloc and
successfully executes this line:

p->tortmp_resample_context = av_audio_resample_init(
                                        1, 1,
                                        p->rtmpinputrate,
p->astinputrate,
                                        SAMPLE_FMT_S16, SAMPLE_FMT_S16,
                                        16, 10, 1, 0.8);


But then fails on this line:

p->fromrtmp_resample_context = av_audio_resample_init(
                                        1, 1,
                                        p->astinputrate,
p->rtmpinputrate,
                                        SAMPLE_FMT_S16, SAMPLE_FMT_S16,
                                        16, 10, 1, 0.8);

Giving either the:
[pcm_s16le @ 0x9aa1b60]codec type or id mismatches
Segmentation fault

or this error:
[nellymoser @ 0x9e276d0]codec type or id mismatches
Segmentation fault

depending on whether I do what vmanthei suggested...

Does that provide any good clues as to my issue? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-12-11 13:25 gpulier        Note Added: 0115154                          
======================================================================




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