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