[svn-commits] phsultan: branch phsultan/rtmp-support r272521 - in /team/phsultan/rtmp-suppo...

[SVN commits to the Digium repositories]

Author: phsultan
Date: Fri Jun 25 07:52:32 2010
New Revision: 272521

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=272521
Log:
Update doc file and configuration example

Modified:
    team/phsultan/rtmp-support/configs/rtmp.conf.sample
    team/phsultan/rtmp-support/doc/rtmp.txt

Modified: team/phsultan/rtmp-support/configs/rtmp.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/phsultan/rtmp-support/configs/rtmp.conf.sample?view=diff&rev=272521&r1=272520&r2=272521
==============================================================================
--- team/phsultan/rtmp-support/configs/rtmp.conf.sample (original)
+++ team/phsultan/rtmp-support/configs/rtmp.conf.sample Fri Jun 25 07:52:32 2010
@@ -16,5 +16,8 @@
 application=oflaDemo
 ;
 ; Based on these parameters, Asterisk will try to establish a connection to
-; the following URL : rtmp://12.34.56.78:1935/applicationName
+; the following URL : rtmp://127.0.0.1:1935/oflaDemo
 ;
+log=yes				; enable librtmp debugging
+rtmplogfile=logfile.txt		; defaults to /tmp/rtmplog.txt
+loglevel=4			; 0 to 6

Modified: team/phsultan/rtmp-support/doc/rtmp.txt
URL: http://svnview.digium.com/svn/asterisk/team/phsultan/rtmp-support/doc/rtmp.txt?view=diff&rev=272521&r1=272520&r2=272521
==============================================================================
--- team/phsultan/rtmp-support/doc/rtmp.txt (original)
+++ team/phsultan/rtmp-support/doc/rtmp.txt Fri Jun 25 07:52:32 2010
@@ -9,13 +9,8 @@
 you can connect Asterisk to an RTMP server such as Red5 or FMS (Flash Media
 Server), and publish or received streams over the connection.
 
-
 Limitations
 -----------
-
-Simulatenous streams :
- Up to 12 simultaneous streams can be published or received to/from the RTMP
- server. This limitation is expected to be overcome soon.
 
 Media support :
  Audio only is supported at the moment.
@@ -27,6 +22,8 @@
 FFMPEG's libavcodec, to transcode and resample audio packets of the proprietary
 codec implemented in Flash clients (Nellymoser's asao).
 
+The RTMP implementation of Asterisk is based on librtmp :
+http://rtmpdump.mplayerhq.hu/
 
 Examples
 --------
@@ -39,10 +36,3 @@
 - read a stream named 'readstream' from the RTMP server, which is expected to
   be published by another Flash client or stored by the RTMP server as a file.
 
-exten => 1234,1,Dial(RTMP/writestream/readstream/2);
-
-The optinal argument '2' indicates Asterisk to start reading two streams from 
-the server, named 'readstream-0' and 'readstream-1', while publishing the 
-stream named 'writestream'. This can be useful if you have a FLEX + Red5/FMS
-conferencing application that you want the terminals managed by Asterisk to 
-enter.