Peter, Here is my rtmp.conf.xml: <configuration name="rtmp.conf" description="RTMP Endpoint"> <profiles> <profile name="default"> <settings> <param name="bind-address" value="0.0.0.0:1935" /> <param name="context" value="public" /> <param name="dialplan" value="XML" /> <!-- If this is set to true, no unauthenticated inbound calls will be allowed --> <param name="auth-calls" value="true" /> <!-- How much time should the clients buffer the media stream (miliseconds) --> <param name="buffer-len" value="50" /> <!-- Sets the maximum size of outbound RTMP chunks --> <param name="chunksize" value="512" /> </settings> </profile> </profiles> </configuration> 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. Jack On 8/7/2012 1:11 AM, Peter Steinbach wrote:
Hello Jack,
here are some answers to your your questions:
Make sure you use a fully qualified username not just the extension 1001@xxx.xxx.xxx.xxx >the XXX would be the IP of your FreeSwitch Server. I am still in the "connect" state, I am not logging in yet. The webpage tells me "Connecting... " and waits forever
did you configure your rtmp.conf.xml and have a matching context in 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 =
"rtmp://xxx.xxx.xxx.xxx/phone"; var rtmpIP="xxx.xxx.xxx.xxx"; 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:
Hi Peter, Make sure you use a fully qualified username not just the extension 1001@xxx.xxx.xxx.xxx 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:
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?
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