[asterisk-bugs] [Asterisk 0015484]: [branch] RTMP support in Asterisk
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Sep 11 03:51:27 CDT 2009
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=15484
======================================================================
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: 1.6.x Version Tracker
Asterisk Version: SVN
Regression: No
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: 2009-09-11 03:51 CDT
======================================================================
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
======================================================================
----------------------------------------------------------------------
(0110566) adamlis (reporter) - 2009-09-11 03:51
https://issues.asterisk.org/view.php?id=15484#c110566
----------------------------------------------------------------------
I've done checkout of repo this week.
1st issue: configuration variable global->port is not used in connection -
it is parsed and assigned to variable "port", but network connection is
done to constant value not this variable "port"
2nd issue: I'm getting "Floating point exception" on below environment:
Debian GNU/Linux i386/i686 Lenny 5.0.2 + mixed packages from Debian
Squeeze (testing).
Basically all environment and packages are taken from Debian Stable Lenny,
but:
1) ffmpeg packages:
||/ Name Version
Description
+++-======================================-======================================-============================================================================================
ii libavcodec-dev 4:0.5+svn20090706-2
development files for libavcodec
ii libavcodec52 4:0.5+svn20090706-2
ffmpeg codec library
ii libavutil-dev 4:0.5+svn20090706-2
development files for libavutil
ii libavutil49 4:0.5+svn20090706-2
ffmpeg utility library
ii libdirac-encoder0 1.0.2-2
open and royalty free high quality codec - encoder library
ii libgsm1 1.0.13-1
Shared libraries for GSM speech compressor
ii liboil0.3 0.3.16-1
Library of Optimized Inner Loops
ii libopenjpeg2 1.3+dfsg-4
JPEG 2000 image compression/decompression library
ii libschroedinger-1.0-0 1.0.7-2
library for encoding/decoding of Dirac video streams
ii libspeex1 1.2~rc1-1
The Speex codec runtime library
2) speex packages:
ii libspeex-dev 1.2~rc1-1
The Speex codec library development files
ii libspeexdsp-dev 1.2~rc1-1
The Speex extended library development files
ii libspeexdsp1 1.2~rc1-1
The Speex extended runtime library
ii speex 1.2~rc1-1
The Speex codec command line tools
Other stuff (gcc, g++, cpp) is taken from Debian Lenny stable.
Asterisk that has been taken from your SVN compile without errors nor
warnings.
Modules chan_rtmp connects to my FMS2 server (Red5 too).
Asterisk is configured using "make samples" + rtmp.conf adjusted to our
server location/port/application.
Then I'm trying to redirect simple exten do RTMP:
[greeting]
exten => 400,1,Answer
exten => 400,n(hihi),BackGround(demo-congrats) ; Play a congratulatory
message
exten => 400,n,Wait(5)
exten => 400,n,HangUp()
By calling from Asterisk CLI:
localhost*CLI> originate RTMP/NetStream/NetStream extension 400 at greeting
I see channels are being registered on FMS2 site - there is kind of system
log of FMS available.
Then output of this 'originate' command:
localhost*CLI> originate RTMP/NetStream/NetStream extension 400 at greeting
Sending createStream request for stream with id 1.000000
Sending createStream request for stream with id 2.000000
[Sep 11 09:03:46] WARNING[11707]: chan_rtmp.c:2798 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 NetStream
[Sep 11 09:03:46] WARNING[11707]: chan_rtmp.c:2798 amf_get_type: Unknown
type 0
Received RTMP message from server :
result : 2.000000
level : N/A
code : N/A
description : N/A
[Sep 11 09:03:46] NOTICE[11707]: chan_rtmp.c:2504
rtmp_handle_connection_message: readstream_index : -0
[Sep 11 09:03:46] NOTICE[11707]: chan_rtmp.c:2509
rtmp_handle_connection_message: readstream_name : NetStream
Sending play request for stream with id 2 and name NetStream
-- Executing [400 at greeting:1] Answer("RTMP/1", "") in new stack
-- Executing [400 at greeting:2] BackGround("RTMP/1", "demo-congrats") in
new stack
-- <RTMP/1> Playing 'demo-congrats.gsm' (language '')
Handling PING message (ping type = 0)
Handling CHUNKSIZE message. Chunk size changed from 128 to 128
Handling PING message (ping type = 0)
Handling CHUNKSIZE message. Chunk size changed from 128 to 441
localhost*CLI> sbin/safe_asterisk: line 152: 11693 Floating point
exception(core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f
${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
Asterisk ended with exit status 136
Asterisk exited on signal EXITSTATUS-128.
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk ending (0).
o2oadmin at localhost:/srv/asterisk$ Automatically restarting Asterisk.
***
Well - seems it does not understand part of FMS2 messages - but same
problem on Red5.
Could anyone help me investigating this problem?
Issue History
Date Modified Username Field Change
======================================================================
2009-09-11 03:51 adamlis Note Added: 0110566
======================================================================
More information about the asterisk-bugs
mailing list