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Peter,<br>
Here is my rtmp.conf.xml:<br>
<br>
<configuration name="rtmp.conf" description="RTMP Endpoint"><br>
<profiles><br>
<profile name="default"><br>
<settings><br>
<param name="bind-address" value="0.0.0.0:1935" /><br>
<big> <param name="context" value="public" /><br>
</big> <param name="dialplan" value="XML" /><br>
<!-- If this is set to true, no unauthenticated
inbound calls will be allowed --><br>
<param name="auth-calls" value="true" /><br>
<!-- How much time should the clients buffer the
media stream (miliseconds) --><br>
<param name="buffer-len" value="50" /><br>
<!-- Sets the maximum size of outbound RTMP chunks
--><br>
<param name="chunksize" value="512" /><br>
</settings><br>
</profile><br>
</profiles><br>
</configuration><br>
<br>
It is important to specify the context and then make sure you have a
condition in your corresponding dial plan that will catch your call.<br>
<br>
Jack<br>
<br>
<br>
<br>
<div class="moz-cite-prefix">On 8/7/2012 1:11 AM, Peter Steinbach
wrote:<br>
</div>
<blockquote cite="mid:5020CDB9.3060102@telefaks.de" type="cite">
<pre wrap="">Hello Jack,
here are some answers to your your questions:
</pre>
<blockquote type="cite">
<pre wrap="">Make sure you use a fully qualified username not just the extension
<a class="moz-txt-link-abbreviated" href="mailto:1001@xxx.xxx.xxx.xxx">1001@xxx.xxx.xxx.xxx</a> >the XXX would be the IP of your FreeSwitch Server.
</pre>
</blockquote>
<pre wrap="">I am still in the "connect" state, I am not logging in yet. The webpage
tells me "Connecting... " and waits forever
</pre>
<blockquote type="cite">
<pre wrap="">did you configure your rtmp.conf.xml and have a matching context in
</pre>
</blockquote>
<pre wrap="">your dial plan?
I am still in the "connect" state
In the web page make sure these two vars are set to YOUR Freeswitch
server ip: var rtmpIPURL =
</pre>
<blockquote type="cite">
<pre wrap="">"rtmp://xxx.xxx.xxx.xxx/phone"; var rtmpIP="xxx.xxx.xxx.xxx";
</pre>
</blockquote>
<pre wrap="">I did not have a rtmpIP var, so I added it to the code, but no change.
In fact as I can see while grepping the network traffic, the Flex app
and Freeswicth are negociating their protocol and freeswitch logs a
"Sent connect reply", but this is very diferent from what is sent from
the conference.freeswitch.org server.
Best regards
Peter
On 08/07/12 06:26, Jack wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Hi Peter,
Make sure you use a fully qualified username not just the extension
<a class="moz-txt-link-abbreviated" href="mailto:1001@xxx.xxx.xxx.xxx">1001@xxx.xxx.xxx.xxx</a>
the XXX would be the IP of your FreeSwitch Server.
did you configure your rtmp.conf.xml and have a matching context in your
dial plan?
In the web page make sure these two vars are set to YOUR Freeswitch
server ip:
var rtmpIPURL = "rtmp://xxx.xxx.xxx.xxx/phone";
var rtmpIP="xxx.xxx.xxx.xxx";
jack
On 8/6/2012 5:44 PM, Peter Steinbach wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Hello,
today I tried the flex client with the mod_rtmp implementation.
On a brandnew freeswitch I installed mod_rtmp and copied the flex
direrectory to a web server and loaded the web page.
However the flex client does not connect:
Here's my freeswitch log
2012-08-07 02:32:03.585671 [NOTICE] mod_rtmp.c:744 New RTMP session
[f35a192c-0d02-4320-89e9-775a3573ee25]
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:702 Sent handshake response
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:727 Done with handshake
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14
ts=0 stream_id=0x0] len=342
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE
for connect
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1
stream_id=0x0] len=4
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5
stream_id=0x0] len=4
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6
stream_id=0x0] len=5
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4
stream_id=0x0] len=6
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
stream_id=0x0] len=201
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
stream_id=0x0] len=61
2012-08-07 02:32:03.685670 [NOTICE] rtmp_sig.c:121 Sent connect reply
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5
ts=12247364 stream_id=0x0] len=4
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:945 Set window size: 131072 bytes
If I change the rtmp_url to 'rtmp://conference.freeswitch.org/phone',
flex does connect.
But in my freeswitch this fails. I have traced the network traffic and I
can see that there is information flow between freeswitch and the client
on connect request.
Somebody has an idea where to look further?
</pre>
</blockquote>
<pre wrap="">
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</pre>
</blockquote>
<pre wrap="">
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
<a class="moz-txt-link-abbreviated" href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://www.freeswitchsolutions.com">http://www.freeswitchsolutions.com</a>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
<a class="moz-txt-link-freetext" href="http://www.cudatel.com">http://www.cudatel.com</a>
Official FreeSWITCH Sites
<a class="moz-txt-link-freetext" href="http://www.freeswitch.org">http://www.freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://wiki.freeswitch.org">http://wiki.freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://www.cluecon.com">http://www.cluecon.com</a>
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FreeSWITCH-users mailing list
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</pre>
</blockquote>
<br>
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