[asterisk-bugs] [Asterisk 0015484]: [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:             1.8
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:              2010-03-07 17:58 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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---------------------------------------------------------------------- 
 (0119073) alerios (reporter) - 2010-03-07 17:58
 https://issues.asterisk.org/view.php?id=15484#c119073 
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I've got the readstream(prueba1) working without noise using shimasaki
patch, but I cannot hear the writestream(prueba2) on publisher.swf. I see
no errors on CLI with debug level 3:

    -- Executing [456 at real-out-alerios:2] Dial("SIP/alerios-00000001",
"RTMP/prueba2/prueba1") in new stack
Sending createStream request for stream with id 1.000000
[100307-184112] WARNING[2958]: chan_rtmp.c:2796 amf_get_type: Unknown type
0
Received RTMP message from server :
         result       : 1.000000
         level        : N/A
         code         : N/A
         description  : N/A
Sending publish request for stream with id 1 and name prueba2
Sending createStream request for stream with id 2.000000
    -- Called prueba2/prueba1
    -- RTMP/1 answered SIP/alerios-00000001
[100307-184112] WARNING[2958]: chan_rtmp.c:2796 amf_get_type: Unknown type
0
Received RTMP message from server :
         result       : 2.000000
         level        : N/A
         code         : N/A
         description  : N/A
[100307-184112] NOTICE[2958]: chan_rtmp.c:2502
rtmp_handle_connection_message: readstream_index : -0
[100307-184112] NOTICE[2958]: chan_rtmp.c:2507
rtmp_handle_connection_message: readstream_name : prueba1
Sending play request for stream with id 2 and name prueba1
Handling PING message (ping type = 0)
Unknown system message with type 3
[100307-184117] NOTICE[2958]: chan_rtmp.c:2647 rtmp_handle_audio_packet:
Changed incoming sample rate from 11000 Hz to 22000 Hz
Unknown system message with type 0
Unknown system message with type 0


The last messages keeps repeating during the whole call.

I'm using Red5 and debian/unstable:

ffmpeg               4:0.5.1-1
libavcodec52         4:0.5.1-1 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-03-07 17:58 alerios        Note Added: 0119073                          
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